git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/video_coding/codecs/OWNERS

@@ -0,0 +1,4 @@
+holmer@google.com
+mikhal@google.com
+marpan@google.com
+hlundin@google.com

modules/video_coding/codecs/i420/main/interface/i420.h

@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
+
+#include "video_codec_interface.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class I420Encoder : public VideoEncoder
+{
+public:
+
+    I420Encoder();
+
+    virtual ~I420Encoder();
+
+// Initialize the encoder with the information from the VideoCodec
+//
+// Input:
+//          - codecSettings     : Codec settings
+//          - numberOfCores     : Number of cores available for the encoder
+//          - maxPayloadSize    : The maximum size each payload is allowed
+//                                to have. Usually MTU - overhead.
+//
+// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+//                                <0 - Error
+    virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/, WebRtc_UWord32 /*maxPayloadSize*/);
+
+// "Encode" an I420 image (as a part of a video stream). The encoded image
+// will be returned to the user via the encode complete callback.
+//
+// Input:
+//          - inputImage        : Image to be encoded
+//          - codecSpecificInfo : Pointer to codec specific data
+//          - frameType         : Frame type to be sent (Key /Delta) .
+//
+// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+//                                <0 - Error
+    virtual WebRtc_Word32 Encode(const RawImage& inputImage, const void* /*codecSpecificInfo*/, VideoFrameType /*frameType*/);
+
+// Register an encode complete callback object.
+//
+// Input:
+//          - callback         : Callback object which handles encoded images.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
+
+// Free encoder memory.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Release();
+
+// Reset encoder state and prepare for a new call.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+//                               <0 - Error
+    virtual WebRtc_Word32 Reset();
+
+    virtual WebRtc_Word32 SetRates(WebRtc_UWord32 /*newBitRate*/, WebRtc_UWord32 /*frameRate*/) {return WEBRTC_VIDEO_CODEC_OK;}
+
+    virtual WebRtc_Word32 SetPacketLoss(WebRtc_UWord32 /*packetLoss*/){return WEBRTC_VIDEO_CODEC_OK;};
+
+    virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/){return WEBRTC_VIDEO_CODEC_OK;};
+
+// Get version number for the codec.
+//
+// Input:
+//      - version       : Pointer to allocated char buffer.
+//      - length        : Length of provided char buffer.
+//
+// Output:
+//      - version       : Version number string written to char buffer.
+//
+// Return value         : >0 - Length of written string.
+//                        <0 - Error
+    static WebRtc_Word32 VersionStatic(WebRtc_Word8 *version, WebRtc_Word32 length);
+    virtual WebRtc_Word32 Version(WebRtc_Word8 *version, WebRtc_Word32 length) const;
+
+private:
+    bool                     _inited;
+    EncodedImage             _encodedImage;
+    EncodedImageCallback*    _encodedCompleteCallback;
+
+}; // end of WebRtcI420DEncoder class
+
+class I420Decoder : public VideoDecoder
+{
+public:
+
+    I420Decoder();
+
+    virtual ~I420Decoder();
+
+// Initialize the decoder.
+// The user must notify the codec of width and height values.
+//
+// Return value         :  WEBRTC_VIDEO_CODEC_OK.
+//                        <0 - Errors
+    virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/);
+
+    virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/){return WEBRTC_VIDEO_CODEC_OK;};
+
+// Decode encoded image (as a part of a video stream). The decoded image
+// will be returned to the user through the decode complete callback.
+//
+// Input:
+//          - inputImage        : Encoded image to be decoded
+//          - missingFrames     : True if one or more frames have been lost
+//                                since the previous decode call.
+//          - codecSpecificInfo : pointer to specific codec data
+//          - renderTimeMs      : Render time in Ms
+//
+// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+//                                 <0 - Error
+    virtual WebRtc_Word32 Decode(const EncodedImage& inputImage, bool missingFrames,
+                                const void* /*codecSpecificInfo */, WebRtc_Word64 /*renderTimeMs*/);
+
+// Register a decode complete callback object.
+//
+// Input:
+//          - callback         : Callback object which handles decoded images.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
+
+// Free decoder memory.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK
+//                                  <0 - Error
+    virtual WebRtc_Word32 Release();
+
+// Reset decoder state and prepare for a new call.
+//
+// Return value         :  WEBRTC_VIDEO_CODEC_OK.
+//                          <0 - Error
+    virtual WebRtc_Word32 Reset();
+
+private:
+
+    RawImage                    _decodedImage;
+    WebRtc_Word32               _width;
+    WebRtc_Word32               _height;
+    bool                        _inited;
+    DecodedImageCallback*       _decodeCompleteCallback;
+
+
+}; // end of WebRtcI420Decoder class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_

modules/video_coding/codecs/i420/main/source/i420.cc

@@ -0,0 +1,279 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "i420.h"
+#include <string.h>
+
+namespace webrtc
+{
+
+I420Encoder::I420Encoder():
+_inited(false),
+_encodedImage(),
+_encodedCompleteCallback(NULL)
+{
+     //
+}
+
+I420Encoder::~I420Encoder()
+{
+    _inited = false;
+    if (_encodedImage._buffer != NULL)
+    {
+        delete [] _encodedImage._buffer;
+        _encodedImage._buffer = NULL;
+    }
+}
+
+WebRtc_Word32
+I420Encoder::VersionStatic(WebRtc_Word8* version, WebRtc_Word32 length)
+{
+    const WebRtc_Word8* str= "I420 version 1.1.0\n";
+    WebRtc_Word32 verLen = (WebRtc_Word32)strlen(str);
+    if(verLen > length)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    strncpy(version, str,length);
+    return verLen;
+}
+
+WebRtc_Word32
+I420Encoder::Version(WebRtc_Word8 *version, WebRtc_Word32 length) const
+{
+    return VersionStatic(version, length);
+}
+
+
+
+WebRtc_Word32
+I420Encoder::Release()
+{
+    // should allocate an encoded frame and then release it here, for that we actaully need an init flag
+    if (_encodedImage._buffer != NULL)
+    {
+        delete [] _encodedImage._buffer;
+        _encodedImage._buffer = NULL;
+    }
+    _inited = false;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Encoder::Reset()
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    return WEBRTC_VIDEO_CODEC_OK;
+
+}
+
+WebRtc_Word32
+I420Encoder::InitEncode(const VideoCodec* codecSettings,
+                              WebRtc_Word32 /*numberOfCores*/,
+                              WebRtc_UWord32 /*maxPayloadSize */)
+{
+    if (codecSettings == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (codecSettings->width < 1 || codecSettings->height < 1)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+
+    // allocating encoded memory
+
+    if (_encodedImage._buffer != NULL)
+    {
+        delete [] _encodedImage._buffer;
+        _encodedImage._buffer = NULL;
+        _encodedImage._size = 0;
+    }
+    const WebRtc_UWord32 newSize = (3 * codecSettings->width *
+                                      codecSettings->height) >> 1;
+    WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+    if (newBuffer == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_MEMORY;
+    }
+    _encodedImage._size = newSize;
+    _encodedImage._buffer = newBuffer;
+
+    // if no memory allocation, no point to init
+    _inited = true;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+
+WebRtc_Word32
+I420Encoder::Encode(const RawImage& inputImage, const void* /*codecSpecificInfo*/, VideoFrameType /*frameTypes*/)
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (_encodedCompleteCallback == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+
+    _encodedImage._frameType = kKeyFrame; // no coding
+    _encodedImage._timeStamp = inputImage._timeStamp;
+    _encodedImage._encodedHeight = inputImage._height;
+    _encodedImage._encodedWidth = inputImage._width;
+    if (inputImage._length > _encodedImage._size)
+    {
+
+        // allocating encoded memory
+        if (_encodedImage._buffer != NULL)
+        {
+            delete [] _encodedImage._buffer;
+            _encodedImage._buffer = NULL;
+            _encodedImage._size = 0;
+        }
+        const WebRtc_UWord32 newSize = (3 * _encodedImage._encodedWidth * _encodedImage._encodedHeight) >> 1;
+        WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+        if (newBuffer == NULL)
+        {
+            return WEBRTC_VIDEO_CODEC_MEMORY;
+        }
+        _encodedImage._size = newSize;
+        _encodedImage._buffer = newBuffer;
+    }
+    memcpy(_encodedImage._buffer, inputImage._buffer, inputImage._length);
+    _encodedImage._length = inputImage._length;
+    _encodedCompleteCallback->Encoded(_encodedImage);
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+WebRtc_Word32
+I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
+{
+    _encodedCompleteCallback = callback;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+I420Decoder::I420Decoder():
+_decodedImage(),
+_width(0),
+_height(0),
+_inited(false),
+_decodeCompleteCallback(NULL)
+{
+    //
+}
+
+I420Decoder::~I420Decoder()
+{
+    Release();
+}
+
+WebRtc_Word32
+I420Decoder::Reset()
+{
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+
+WebRtc_Word32
+I420Decoder::InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores */)
+{
+    if (codecSettings == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    else if (codecSettings->width < 1 || codecSettings->height < 1)
+    {
+         return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    _width = codecSettings->width;
+    _height = codecSettings->height;
+    _inited = true;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/, const void* /*codecSpecificInfo*/, WebRtc_Word64 /*renderTimeMs*/)
+{
+    if (inputImage._buffer == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (_decodeCompleteCallback == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (inputImage._length <= 0)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (!_inited)
+    {
+       return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+
+    //Allocate memory for decoded image
+
+    if (_decodedImage._buffer != NULL)
+    {
+        delete [] _decodedImage._buffer;
+        _decodedImage._buffer = NULL;
+        _decodedImage._size = 0;
+    }
+    if (_decodedImage._buffer == NULL)
+    {
+        const WebRtc_UWord32 newSize = (3*_width*_height) >> 1;
+        WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
+        if (newBuffer == NULL)
+        {
+            return WEBRTC_VIDEO_CODEC_MEMORY;
+        }
+        _decodedImage._size = newSize;
+        _decodedImage._buffer = newBuffer;
+    }
+
+    // Set decoded image parameters
+    _decodedImage._height = _height;
+    _decodedImage._width = _width;
+    _decodedImage._timeStamp = inputImage._timeStamp;
+    memcpy(_decodedImage._buffer, inputImage._buffer, inputImage._length);
+    _decodedImage._length = inputImage._length;
+    //_decodedImage._buffer = inputImage._buffer;
+
+    _decodeCompleteCallback->Decoded(_decodedImage);
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::RegisterDecodeCompleteCallback(DecodedImageCallback* callback)
+{
+    _decodeCompleteCallback = callback;
+        return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+I420Decoder::Release()
+{
+    if (_decodedImage._buffer != NULL)
+    {
+        delete [] _decodedImage._buffer;
+        _decodedImage._buffer = NULL;
+    }
+    _inited = false;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+}

modules/video_coding/codecs/i420/main/source/i420.gyp

@@ -0,0 +1,37 @@
+# Copyright (c) 2009 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+  'includes': [
+    '../../../../../../common_settings.gypi', # Common settings
+  ],
+  'targets': [
+    {
+      'target_name': 'webrtc_i420',
+      'type': '<(library)',
+      'dependencies': [
+        '../../../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
+      ],
+      'include_dirs': [
+        '../interface',
+        '../../../interface',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '../interface',
+        ],
+      },
+      'sources': [
+        '../interface/i420.h',
+        'i420.cc',
+      ],
+    },
+  ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:

modules/video_coding/codecs/interface/video_codec_interface.h

@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VIDEO_CODEC_INTERFACE_H
+#define VIDEO_CODEC_INTERFACE_H
+
+#include "common_types.h"
+#include "typedefs.h"
+#include "video_image.h"
+#include "video_error_codes.h"
+
+namespace webrtc
+{
+
+class RTPFragmentationHeader; // forward declaration
+
+struct CodecSpecificInfoVP8
+{
+    bool             hasReceivedSLI;
+    WebRtc_UWord8    pictureIdSLI;
+    bool             hasReceivedRPSI;
+    WebRtc_UWord64   pictureIdRPSI;
+};
+
+union CodecSpecificInfoUnion
+{
+    CodecSpecificInfoVP8       VP8;
+};
+
+struct CodecSpecificInfo
+{
+    VideoCodecType   codecType;
+    CodecSpecificInfoUnion codecSpecific;
+};
+
+class EncodedImageCallback
+{
+public:
+    virtual ~EncodedImageCallback() {};
+
+    // Callback function which is called when an image has been encoded.
+    //
+    // Input:
+    //          - encodedImage         : The encoded image
+    //
+    // Return value                    : > 0,   signals to the caller that one or more future frames
+    //                                          should be dropped to keep bit rate or frame rate.
+    //                                   = 0,   if OK.
+    //                                   < 0,   on error.
+    virtual WebRtc_Word32 Encoded(EncodedImage& encodedImage,
+                                  const void* codecSpecificInfo = NULL,
+                                  const RTPFragmentationHeader* fragmentation = NULL) = 0;
+};
+
+class VideoEncoder
+{
+public:
+    virtual ~VideoEncoder() {};
+
+    // Get the encoder version.
+    //
+    // Input:
+    //          - length            : Length of the version buffer.
+    //
+    // Output:
+    //          - version           : Buffer where the version string will be written.
+    //
+    // Return value                 : Number of bytes written to the version buffer.
+    //                                < 0 on failure.
+    virtual WebRtc_Word32 Version(WebRtc_Word8 *version, WebRtc_Word32 length) const = 0;
+
+    // Initialize the encoder with the information from the VideoCodec.
+    //
+    // Input:
+    //          - codecSettings     : Codec settings
+    //          - numberOfCores     : Number of cores available for the encoder
+    //          - maxPayloadSize    : The maximum size each payload is allowed
+    //                                to have. Usually MTU - overhead.
+    //
+    // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores, WebRtc_UWord32 maxPayloadSize) = 0;
+
+    // Encode an I420 image (as a part of a video stream). The encoded image
+    // will be returned to the user through the encode complete callback.
+    //
+    // Input:
+    //          - inputImage        : Image to be encoded
+    //          - codecSpecificInfo : Pointer to codec specific data
+    //          - frameType         : The frame type to encode
+    //
+    // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Encode(const RawImage& inputImage,
+                                 const void* codecSpecificInfo = NULL,
+                                 VideoFrameType frameType = kDeltaFrame) = 0;
+
+    // Register an encode complete callback object.
+    //
+    // Input:
+    //          - callback         : Callback object which handles encoded images.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback) = 0;
+
+    // Free encoder memory.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Release() = 0;
+
+    // Reset encoder state and prepare for a new call.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Reset() = 0;
+
+    // Inform the encoder about the packet loss and round trip time on the network
+    // used to decide the best pattern and signaling.
+    //
+    //          - packetLoss       : Fraction lost
+    //                               (loss rate in percent = 100 * packetLoss / 255)
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 SetPacketLoss(WebRtc_UWord32 packetLoss) = 0;
+
+    // Inform the encoder about the new target bit rate.
+    //
+    //          - newBitRate       : New target bit rate
+    //          - frameRate        : The target frame rate
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate) = 0;
+
+    // Use this function to enable or disable periodic key frames. Can be useful for codecs
+    // which have other ways of stopping error propagation.
+    //
+    //          - enable           : Enable or disable periodic key frames
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable) { return WEBRTC_VIDEO_CODEC_ERROR; }
+
+    // Codec configuration data to send out-of-band, i.e. in SIP call setup
+    //
+    //          - buffer           : Buffer pointer to where the configuration data
+    //                               should be stored
+    //          - size             : The size of the buffer in bytes
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
+};
+
+class DecodedImageCallback
+{
+public:
+    virtual ~DecodedImageCallback() {};
+
+    // Callback function which is called when an image has been decoded.
+    //
+    // Input:
+    //          - decodedImage         : The decoded image
+    //
+    // Return value                    : 0 if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Decoded(RawImage& decodedImage) = 0;
+
+    virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
+
+    virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId) {return -1;}
+};
+
+class VideoDecoder
+{
+public:
+    virtual ~VideoDecoder() {};
+
+    // Initialize the decoder with the information from the VideoCodec.
+    //
+    // Input:
+    //          - inst              : Codec settings
+    //          - numberOfCores     : Number of cores available for the decoder
+    //
+    // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores) = 0;
+
+    // Decode encoded image (as a part of a video stream). The decoded image
+    // will be returned to the user through the decode complete callback.
+    //
+    // Input:
+    //          - inputImage        : Encoded image to be decoded
+    //          - missingFrames     : True if one or more frames have been lost
+    //                                since the previous decode call.
+    //          - codecSpecificInfo : Pointer to codec specific data
+    //          - renderTimeMs      : System time to render in milliseconds. Only
+    //                                used by decoders with internal rendering.
+    //
+    // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Decode(const EncodedImage& inputImage, bool missingFrames, const void* codecSpecificInfo = NULL, WebRtc_Word64 renderTimeMs = -1) = 0;
+
+    // Register an decode complete callback object.
+    //
+    // Input:
+    //          - callback         : Callback object which handles decoded images.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback) = 0;
+
+    // Free decoder memory.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Release() = 0;
+
+    // Reset decoder state and prepare for a new call.
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Reset() = 0;
+
+    // Codec configuration data sent out-of-band, i.e. in SIP call setup
+    //
+    // Input/Output:
+    //          - buffer           : Buffer pointer to the configuration data
+    //          - size             : The size of the configuration data in
+    //                               bytes
+    //
+    // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
+
+    // Create a copy of the codec and its internal state.
+    //
+    // Return value                : A copy of the instance if OK, NULL otherwise.
+    virtual VideoDecoder* Copy() { return NULL; }
+};
+
+} // namespace webrtc
+
+#endif // VIDEO_CODEC_INTERFACE_H

modules/video_coding/codecs/interface/video_error_codes.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VIDEO_ERROR_CODES_H
+#define VIDEO_ERROR_CODES_H
+
+// NOTE: in sync with video_coding_module_defines.h
+
+// Define return values
+
+#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
+#define WEBRTC_VIDEO_CODEC_OK 0
+#define WEBRTC_VIDEO_CODEC_ERROR -1
+#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
+#define WEBRTC_VIDEO_CODEC_MEMORY -3
+#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
+#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
+#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
+#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
+#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
+
+#endif // VIDEO_ERROR_CODES_H

modules/video_coding/codecs/interface/video_image.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VIDEO_IMAGE_H
+#define VIDEO_IMAGE_H
+
+#include "typedefs.h"
+#include <stdlib.h>
+
+namespace webrtc
+{
+
+enum VideoFrameType
+{
+    kKeyFrame = 0,
+    kDeltaFrame = 1,
+    kGoldenFrame = 2,
+    kAltRefFrame = 3,
+    kSkipFrame = 4
+};
+
+class RawImage
+{
+public:
+    RawImage() :    _width(0), _height(0), _timeStamp(0), _buffer(NULL),
+                    _length(0), _size(0) {}
+
+    RawImage(WebRtc_UWord8* buffer, WebRtc_UWord32 length,
+        WebRtc_UWord32 size) :
+                    _width(0), _height(0), _timeStamp(0),
+                    _buffer(buffer), _length(length), _size(size) {}
+
+    WebRtc_UWord32    _width;
+    WebRtc_UWord32    _height;
+    WebRtc_UWord32    _timeStamp;
+    WebRtc_UWord8*    _buffer;
+    WebRtc_UWord32    _length;
+    WebRtc_UWord32    _size;
+};
+
+class EncodedImage
+{
+public:
+    EncodedImage() :
+                     _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
+                     _frameType(kDeltaFrame), _buffer(NULL), _length(0), _size(0),
+                     _completeFrame(false) {}
+
+    EncodedImage(WebRtc_UWord8* buffer,
+                 WebRtc_UWord32 length,
+                 WebRtc_UWord32 size) :
+                     _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
+                     _frameType(kDeltaFrame), _buffer(buffer), _length(length),
+                     _size(size), _completeFrame(false) {}
+
+    WebRtc_UWord32              _encodedWidth;
+    WebRtc_UWord32              _encodedHeight;
+    WebRtc_UWord32              _timeStamp;
+    VideoFrameType             _frameType;
+    WebRtc_UWord8*              _buffer;
+    WebRtc_UWord32              _length;
+    WebRtc_UWord32              _size;
+    bool                        _completeFrame;
+};
+
+} // namespace webrtc
+
+#endif // VIDEO_IMAGE_H

modules/video_coding/codecs/test_framework/benchmark.cc

@@ -0,0 +1,304 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "benchmark.h"
+#include "video_source.h"
+#include "vplib.h"
+#include <vector>
+#include <fstream>
+#include <iostream>
+#include <sstream>
+#include <cassert>
+#if defined(_WIN32)
+    #include <windows.h>
+#endif
+#include "event_wrapper.h"
+#include "video_codec_interface.h"
+
+#define SSIM_CALC 0 // by default, don't compute SSIM
+
+using namespace webrtc;
+
+Benchmark::Benchmark()
+:
+_resultsFileName("../../../../testFiles/benchmark.txt"),
+_codecName("Default"),
+NormalAsyncTest("Benchmark", "Codec benchmark over a range of test cases", 6)
+{
+}
+
+Benchmark::Benchmark(std::string name, std::string description)
+:
+_resultsFileName("../../../../testFiles/benchmark.txt"),
+_codecName("Default"),
+NormalAsyncTest(name, description, 6)
+{
+}
+
+Benchmark::Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName)
+:
+_resultsFileName(resultsFileName),
+_codecName(codecName),
+NormalAsyncTest(name, description, 6)
+{
+}
+
+void
+Benchmark::Perform()
+{
+    std::vector<const VideoSource*> sources;
+    std::vector<const VideoSource*>::iterator it;
+
+    // Configuration --------------------------
+    sources.push_back(new const VideoSource("test/testFiles/foreman_cif.yuv", kCIF));
+    sources.push_back(new const VideoSource("test/testFiles/akiyo_cif.yuv", kCIF));
+
+    const VideoSize size[] = {kQCIF, kCIF};
+    const int frameRate[] = {10, 15, 30};
+    // Specifies the framerates for which to perform a speed test.
+    const bool speedTestMask[] = {false, false, false};
+    const int bitRate[] = {50, 100, 200, 300, 400, 500, 600, 1000};
+    // Determines the number of iterations to perform to arrive at the speed result.
+    enum { kSpeedTestIterations = 10 };
+    // ----------------------------------------
+
+    const int nFrameRates = sizeof(frameRate)/sizeof(*frameRate);
+    assert(sizeof(speedTestMask)/sizeof(*speedTestMask) == nFrameRates);
+    const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
+    int testIterations = 10;
+
+    double psnr[nBitrates];
+    double ssim[nBitrates];
+    double fps[nBitrates];
+    double totalEncodeTime[nBitrates];
+    double totalDecodeTime[nBitrates];
+
+    _results.open(_resultsFileName.c_str(), std::fstream::out);
+    _results << GetMagicStr() << std::endl;
+    _results << _codecName << std::endl;
+
+    for (it = sources.begin() ; it < sources.end(); it++)
+    {
+        for (int i = 0; i < sizeof(size)/sizeof(*size); i++)
+        {
+            for (int j = 0; j < nFrameRates; j++)
+            {
+                std::stringstream ss;
+                std::string strFrameRate;
+                std::string outFileName;
+                ss << frameRate[j];
+                ss >> strFrameRate;
+                outFileName = (*it)->GetFilePath() + "/" + (*it)->GetName() + "_" +
+                    VideoSource::GetSizeString(size[i]) + "_" + strFrameRate + ".yuv";
+
+                _target = new const VideoSource(outFileName, size[i], frameRate[j]);
+                (*it)->Convert(*_target);
+                if (VideoSource::FileExists(outFileName.c_str()))
+                {
+                    _inname = outFileName;
+                }
+                else
+                {
+                    _inname = (*it)->GetFileName();
+                }
+
+                std::cout << (*it)->GetName() << ", " << VideoSource::GetSizeString(size[i])
+                    << ", " << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]:";
+                _results << (*it)->GetName() << "," << VideoSource::GetSizeString(size[i])
+                    << "," << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]";
+
+                if (speedTestMask[j])
+                {
+                    testIterations = kSpeedTestIterations;
+                }
+                else
+                {
+                    testIterations = 1;
+                }
+
+                for (int k = 0; k < nBitrates; k++)
+                {
+                    _bitRate = (bitRate[k]);
+                    double avgFps = 0.0;
+                    totalEncodeTime[k] = 0;
+                    totalDecodeTime[k] = 0;
+
+                    for (int l = 0; l < testIterations; l++)
+                    {
+                        PerformNormalTest();
+                        _appendNext = false;
+
+                        avgFps += _framecnt / (_totalEncodeTime + _totalDecodeTime);
+                        totalEncodeTime[k] += _totalEncodeTime;
+                        totalDecodeTime[k] += _totalDecodeTime;
+
+                    }
+                    avgFps /= testIterations;
+                    totalEncodeTime[k] /= testIterations;
+                    totalDecodeTime[k] /= testIterations;
+
+                    double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
+                    std::cout << " " << actualBitRate;
+                    _results << "," << actualBitRate;
+                    PSNRfromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
+                        _inst.height, &psnr[k]);
+                    if (SSIM_CALC)
+                    {
+                        SSIMfromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
+                        _inst.height, &ssim[k]);
+
+                    }
+                    fps[k] = avgFps;
+                }
+                std::cout << std::endl << "Y-PSNR [dB]:";
+                _results << std::endl << "Y-PSNR [dB]";
+                for (int k = 0; k < nBitrates; k++)
+                {
+                    std::cout << " " << psnr[k];
+                    _results << "," << psnr[k];
+
+                }
+                if (SSIM_CALC)
+                {
+                    std::cout << std::endl << "SSIM: ";
+                    _results << std::endl << "SSIM ";
+                    for (int k = 0; k < nBitrates; k++)
+                    {
+                        std::cout << " " << ssim[k];
+                        _results << "," << ssim[k];
+                    }
+
+                }
+
+                std::cout << std::endl << "Encode Time[ms]:";
+                _results << std::endl << "Encode Time[ms]";
+                for (int k = 0; k < nBitrates; k++)
+                {
+                    std::cout << " " << totalEncodeTime[k];
+                    _results << "," << totalEncodeTime[k];
+
+                }
+
+                std::cout << std::endl << "Decode Time[ms]:";
+                _results << std::endl << "Decode Time[ms]";
+                for (int k = 0; k < nBitrates; k++)
+                {
+                    std::cout << " " << totalDecodeTime[k];
+                    _results << "," << totalDecodeTime[k];
+
+                }
+
+                if (speedTestMask[j])
+                {
+                    std::cout << std::endl << "Speed [fps]:";
+                    _results << std::endl << "Speed [fps]";
+                    for (int k = 0; k < nBitrates; k++)
+                    {
+                        std::cout << " " << static_cast<int>(fps[k] + 0.5);
+                        _results << "," << static_cast<int>(fps[k] + 0.5);
+                    }
+                }
+                std::cout << std::endl << std::endl;
+                _results << std::endl << std::endl;
+
+                delete _target;
+            }
+        }
+        delete *it;
+    }
+    _results.close();
+}
+
+void
+Benchmark::PerformNormalTest()
+{
+    _encoder = GetNewEncoder();
+    _decoder = GetNewDecoder();
+    CodecSettings(_target->GetWidth(), _target->GetHeight(), _target->GetFrameRate(), _bitRate);
+    Setup();
+    EventWrapper* waitEvent = EventWrapper::Create();
+
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _encoder->InitEncode(&_inst, 4, 1440);
+    CodecSpecific_InitBitrate();
+    _decoder->InitDecode(&_inst,1);
+
+    FrameQueue frameQueue;
+    VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+    VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
+    _encoder->RegisterEncodeCompleteCallback(&encCallback);
+    _decoder->RegisterDecodeCompleteCallback(&decCallback);
+
+    SetCodecSpecificParameters();
+
+    _totalEncodeTime = _totalDecodeTime = 0;
+    _totalEncodePipeTime = _totalDecodePipeTime = 0;
+    bool complete = false;
+    _framecnt = 0;
+    _encFrameCnt = 0;
+    _sumEncBytes = 0;
+    _lengthEncFrame = 0;
+    while (!complete)
+    {
+        complete = Encode();
+        if (!frameQueue.Empty() || complete)
+        {
+            while (!frameQueue.Empty())
+            {
+                _frameToDecode = static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+                DoPacketLoss();
+                int ret = Decode();
+                delete _frameToDecode;
+                _frameToDecode = NULL;
+                if (ret < 0)
+                {
+                    fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+                    exit(EXIT_FAILURE);
+                }
+                else if (ret == 0)
+                {
+                    _framecnt++;
+                }
+                else
+                {
+                    fprintf(stderr, "\n\nPositive return value from decode!\n\n");
+                }
+            }
+        }
+        waitEvent->Wait(5);
+    }
+
+    _inputVideoBuffer.Free();
+    //_encodedVideoBuffer.Reset(); ?
+    _encodedVideoBuffer.Free();
+    _decodedVideoBuffer.Free();
+
+    _encoder->Release();
+    _decoder->Release();
+    delete waitEvent;
+    delete _encoder;
+    delete _decoder;
+    Teardown();
+}
+
+void
+Benchmark::CodecSpecific_InitBitrate()
+{
+    if (_bitRate == 0)
+    {
+        _encoder->SetRates(600, _inst.maxFramerate);
+    }
+    else
+    {
+        _encoder->SetRates(_bitRate, _inst.maxFramerate);
+    }
+}
+

modules/video_coding/codecs/test_framework/benchmark.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+
+#include "normal_async_test.h"
+
+class VideoSource;
+
+class Benchmark : public NormalAsyncTest
+{
+public:
+    Benchmark();
+    virtual void Perform();
+
+protected:
+    Benchmark(std::string name, std::string description);
+    Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName);
+    virtual webrtc::VideoEncoder* GetNewEncoder() = 0;
+    virtual webrtc::VideoDecoder* GetNewDecoder() = 0;
+    virtual void PerformNormalTest();
+    virtual void CodecSpecific_InitBitrate();
+    static const char* GetMagicStr() { return "#!benchmark1.0"; }
+
+    const VideoSource* _target;
+    std::string        _resultsFileName;
+    std::ofstream      _results;
+    std::string        _codecName;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
+

modules/video_coding/codecs/test_framework/exportfig.m

@@ -0,0 +1,500 @@
+function exportfig(varargin)
+%EXPORTFIG  Export a figure to Encapsulated Postscript.
+%   EXPORTFIG(H, FILENAME) writes the figure H to FILENAME.  H is
+%   a figure handle and FILENAME is a string that specifies the
+%   name of the output file.
+%
+%   EXPORTFIG(...,PARAM1,VAL1,PARAM2,VAL2,...) specifies
+%   parameters that control various characteristics of the output
+%   file.
+%
+%   Format Paramter:
+%     'Format'  one of the strings 'eps','eps2','jpeg','png','preview'
+%          specifies the output format. Defaults to 'eps'.
+%          The output format 'preview' does not generate an output
+%          file but instead creates a new figure window with a
+%          preview of the exported figure. In this case the
+%          FILENAME parameter is ignored.
+%
+%     'Preview' one of the strings 'none', 'tiff'
+%          specifies a preview for EPS files. Defaults to 'none'.
+%
+%   Size Parameters:
+%     'Width'   a positive scalar
+%          specifies the width in the figure's PaperUnits
+%     'Height'  a positive scalar
+%          specifies the height in the figure's PaperUnits
+%
+%     Specifying only one dimension sets the other dimension
+%     so that the exported aspect ratio is the same as the
+%     figure's current aspect ratio. 
+%     If neither dimension is specified the size defaults to 
+%     the width and height from the figure's PaperPosition. 
+%           
+%   Rendering Parameters:
+%     'Color'     one of the strings 'bw', 'gray', 'cmyk'
+%         'bw'    specifies that lines and text are exported in
+%                 black and all other objects in grayscale
+%         'gray'  specifies that all objects are exported in grayscale
+%         'cmyk'  specifies that all objects are exported in color
+%                 using the CMYK color space
+%     'Renderer'  one of the strings 'painters', 'zbuffer', 'opengl'
+%         specifies the renderer to use
+%     'Resolution'   a positive scalar
+%         specifies the resolution in dots-per-inch.
+%     
+%     The default color setting is 'bw'.
+%
+%   Font Parameters:
+%     'FontMode'     one of the strings 'scaled', 'fixed'
+%     'FontSize'     a positive scalar
+%          in 'scaled' mode multiplies with the font size of each
+%          text object to obtain the exported font size
+%          in 'fixed' mode specifies the font size of all text
+%          objects in points
+%     'FontEncoding' one of the strings 'latin1', 'adobe'
+%          specifies the character encoding of the font
+%
+%     If FontMode is 'scaled' but FontSize is not specified then a
+%     scaling factor is computed from the ratio of the size of the
+%     exported figure to the size of the actual figure. The minimum
+%     font size allowed after scaling is 5 points.
+%     If FontMode is 'fixed' but FontSize is not specified then the
+%     exported font sizes of all text objects is 7 points.
+%
+%     The default 'FontMode' setting is 'scaled'.
+%
+%   Line Width Parameters:
+%     'LineMode'     one of the strings 'scaled', 'fixed'
+%     'LineWidth'    a positive scalar
+%          the semantics of LineMode and LineWidth are exactly the
+%          same as FontMode and FontSize, except that they apply
+%          to line widths instead of font sizes. The minumum line
+%          width allowed after scaling is 0.5 points.
+%          If LineMode is 'fixed' but LineWidth is not specified 
+%          then the exported line width of all line objects is 1
+%          point. 
+%
+%   Examples:
+%     exportfig(gcf,'fig1.eps','height',3);
+%       Exports the current figure to the file named 'fig1.eps' with
+%       a height of 3 inches (assuming the figure's PaperUnits is 
+%       inches) and an aspect ratio the same as the figure's aspect
+%       ratio on screen.
+%
+%     exportfig(gcf, 'fig2.eps', 'FontMode', 'fixed',...
+%                'FontSize', 10, 'color', 'cmyk' );
+%       Exports the current figure to 'fig2.eps' in color with all
+%       text in 10 point fonts. The size of the exported figure is
+%       the figure's PaperPostion width and height.
+
+
+if (nargin < 2)
+  error('Too few input arguments');
+end
+
+% exportfig(H, filename, ...)
+H = varargin{1};
+if ~ishandle(H) | ~strcmp(get(H,'type'), 'figure')
+  error('First argument must be a handle to a figure.');
+end
+filename = varargin{2};
+if ~ischar(filename)
+  error('Second argument must be a string.');
+end
+paramPairs = varargin(3:end);
+
+% Do some validity checking on param-value pairs
+if (rem(length(paramPairs),2) ~= 0)
+  error(['Invalid input syntax. Optional parameters and values' ...
+	 ' must be in pairs.']);
+end
+
+format = 'eps';
+preview = 'none';
+width = -1;
+height = -1;
+color = 'bw';
+fontsize = -1;
+fontmode='scaled';
+linewidth = -1;
+linemode=[];
+fontencoding = 'latin1';
+renderer = [];
+resolution = [];
+
+% Process param-value pairs
+args = {};
+for k = 1:2:length(paramPairs)
+  param = lower(paramPairs{k});
+  if (~ischar(param))
+    error('Optional parameter names must be strings');
+  end
+  value = paramPairs{k+1};
+  
+  switch (param)
+   case 'format'
+    format = value;
+    if (~strcmp(format,{'eps','eps2','jpeg','png','preview'}))
+      error(['Format must be ''eps'', ''eps2'', ''jpeg'', ''png'' or' ...
+	     ' ''preview''.']);
+    end
+   case 'preview'
+    preview = value;
+    if (~strcmp(preview,{'none','tiff'}))
+      error('Preview must be ''none'' or ''tiff''.');
+    end
+   case 'width'
+    width = LocalToNum(value);
+    if(~LocalIsPositiveScalar(width))
+      error('Width must be a numeric scalar > 0');
+    end
+   case 'height'
+    height = LocalToNum(value);
+    if(~LocalIsPositiveScalar(height))
+      error('Height must be a numeric scalar > 0');
+    end
+   case 'color'
+    color = lower(value);
+    if (~strcmp(color,{'bw','gray','cmyk'}))
+      error('Color must be ''bw'', ''gray'' or ''cmyk''.');
+    end
+   case 'fontmode'
+    fontmode = lower(value);
+    if (~strcmp(fontmode,{'scaled','fixed'}))
+      error('FontMode must be ''scaled'' or ''fixed''.');
+    end
+   case 'fontsize'
+    fontsize = LocalToNum(value);
+    if(~LocalIsPositiveScalar(fontsize))
+      error('FontSize must be a numeric scalar > 0');
+    end
+   case 'fontencoding'
+    fontencoding = lower(value);
+    if (~strcmp(fontencoding,{'latin1','adobe'}))
+      error('FontEncoding must be ''latin1'' or ''adobe''.');
+    end
+   case 'linemode'
+    linemode = lower(value);
+    if (~strcmp(linemode,{'scaled','fixed'}))
+      error('LineMode must be ''scaled'' or ''fixed''.');
+    end
+   case 'linewidth'
+    linewidth = LocalToNum(value);
+    if(~LocalIsPositiveScalar(linewidth))
+      error('LineWidth must be a numeric scalar > 0');
+    end
+   case 'renderer'
+    renderer = lower(value);
+    if (~strcmp(renderer,{'painters','zbuffer','opengl'}))
+      error('Renderer must be ''painters'', ''zbuffer'' or ''opengl''.');
+    end
+   case 'resolution'
+    resolution = LocalToNum(value);
+    if ~(isnumeric(value) & (prod(size(value)) == 1) & (value >= 0));
+      error('Resolution must be a numeric scalar >= 0');
+    end
+   otherwise
+    error(['Unrecognized option ' param '.']);
+  end
+end
+
+allLines  = findall(H, 'type', 'line');
+allText   = findall(H, 'type', 'text');
+allAxes   = findall(H, 'type', 'axes');
+allImages = findall(H, 'type', 'image');
+allLights = findall(H, 'type', 'light');
+allPatch  = findall(H, 'type', 'patch');
+allSurf   = findall(H, 'type', 'surface');
+allRect   = findall(H, 'type', 'rectangle');
+allFont   = [allText; allAxes];
+allColor  = [allLines; allText; allAxes; allLights];
+allMarker = [allLines; allPatch; allSurf];
+allEdge   = [allPatch; allSurf];
+allCData  = [allImages; allPatch; allSurf];
+
+old.objs = {};
+old.prop = {};
+old.values = {};
+
+% Process format and preview parameter
+showPreview = strcmp(format,'preview');
+if showPreview
+  format = 'png';
+  filename = [tempName '.png'];
+end
+if strncmp(format,'eps',3) & ~strcmp(preview,'none')
+  args = {args{:}, ['-' preview]};
+end
+
+hadError = 0;
+try
+  % Process size parameters
+  paperPos = get(H, 'PaperPosition');
+  old = LocalPushOldData(old, H, 'PaperPosition', paperPos);
+  figureUnits = get(H, 'Units');
+  set(H, 'Units', get(H,'PaperUnits'));
+  figurePos = get(H, 'Position');
+  aspectRatio = figurePos(3)/figurePos(4);
+  set(H, 'Units', figureUnits);
+  if (width == -1) & (height == -1)
+    width = paperPos(3);
+    height = paperPos(4);
+  elseif (width == -1)
+    width = height * aspectRatio;
+  elseif (height == -1)
+    height = width / aspectRatio;
+  end
+  set(H, 'PaperPosition', [0 0 width height]);
+  paperPosMode = get(H, 'PaperPositionMode');
+  old = LocalPushOldData(old, H, 'PaperPositionMode', paperPosMode);
+  set(H, 'PaperPositionMode', 'manual');
+
+  % Process rendering parameters
+  switch (color)
+   case {'bw', 'gray'}
+    if ~strcmp(color,'bw') & strncmp(format,'eps',3)
+      format = [format 'c'];
+    end
+    args = {args{:}, ['-d' format]};
+
+    %compute and set gray colormap
+    oldcmap = get(H,'Colormap');
+    newgrays = 0.30*oldcmap(:,1) + 0.59*oldcmap(:,2) + 0.11*oldcmap(:,3);
+    newcmap = [newgrays newgrays newgrays];
+    old = LocalPushOldData(old, H, 'Colormap', oldcmap);
+    set(H, 'Colormap', newcmap);
+
+    %compute and set ColorSpec and CData properties
+    old = LocalUpdateColors(allColor, 'color', old);
+    old = LocalUpdateColors(allAxes, 'xcolor', old);
+    old = LocalUpdateColors(allAxes, 'ycolor', old);
+    old = LocalUpdateColors(allAxes, 'zcolor', old);
+    old = LocalUpdateColors(allMarker, 'MarkerEdgeColor', old);
+    old = LocalUpdateColors(allMarker, 'MarkerFaceColor', old);
+    old = LocalUpdateColors(allEdge, 'EdgeColor', old);
+    old = LocalUpdateColors(allEdge, 'FaceColor', old);
+    old = LocalUpdateColors(allCData, 'CData', old);
+    
+   case 'cmyk'
+    if strncmp(format,'eps',3)
+      format = [format 'c'];
+      args = {args{:}, ['-d' format], '-cmyk'};
+    else
+      args = {args{:}, ['-d' format]};
+    end
+   otherwise
+    error('Invalid Color parameter');
+  end
+  if (~isempty(renderer))
+    args = {args{:}, ['-' renderer]};
+  end
+  if (~isempty(resolution)) | ~strncmp(format,'eps',3)
+    if isempty(resolution)
+      resolution = 0;
+    end
+    args = {args{:}, ['-r' int2str(resolution)]};
+  end
+
+  % Process font parameters
+  if (~isempty(fontmode))
+    oldfonts = LocalGetAsCell(allFont,'FontSize');
+    switch (fontmode)
+     case 'fixed'
+      oldfontunits = LocalGetAsCell(allFont,'FontUnits');
+      old = LocalPushOldData(old, allFont, {'FontUnits'}, oldfontunits);
+      set(allFont,'FontUnits','points');
+      if (fontsize == -1)
+	set(allFont,'FontSize',7);
+      else
+	set(allFont,'FontSize',fontsize);
+      end
+     case 'scaled'
+      if (fontsize == -1)
+	wscale = width/figurePos(3);
+	hscale = height/figurePos(4);
+	scale = min(wscale, hscale);
+      else
+	scale = fontsize;
+      end
+      newfonts = LocalScale(oldfonts,scale,5);
+      set(allFont,{'FontSize'},newfonts);
+     otherwise
+      error('Invalid FontMode parameter');
+    end
+    % make sure we push the size after the units
+    old = LocalPushOldData(old, allFont, {'FontSize'}, oldfonts);
+  end
+  if strcmp(fontencoding,'adobe') & strncmp(format,'eps',3)
+    args = {args{:}, '-adobecset'};
+  end
+
+  % Process linewidth parameters
+  if (~isempty(linemode))
+    oldlines = LocalGetAsCell(allMarker,'LineWidth');
+    old = LocalPushOldData(old, allMarker, {'LineWidth'}, oldlines);
+    switch (linemode)
+     case 'fixed'
+      if (linewidth == -1)
+	set(allMarker,'LineWidth',1);
+      else
+	set(allMarker,'LineWidth',linewidth);
+      end
+     case 'scaled'
+      if (linewidth == -1)
+	wscale = width/figurePos(3);
+	hscale = height/figurePos(4);
+	scale = min(wscale, hscale);
+      else
+	scale = linewidth;
+      end
+      newlines = LocalScale(oldlines, scale, 0.5);
+      set(allMarker,{'LineWidth'},newlines);
+     otherwise
+      error('Invalid LineMode parameter');
+    end
+  end
+
+  % Export
+  print(H, filename, args{:});
+
+catch
+  hadError = 1;
+end
+
+% Restore figure settings
+for n=1:length(old.objs)
+  set(old.objs{n}, old.prop{n}, old.values{n});
+end
+
+if hadError
+  error(deblank(lasterr));
+end
+
+% Show preview if requested
+if showPreview
+  X = imread(filename,'png');
+  delete(filename);
+  f = figure( 'Name', 'Preview', ...
+	      'Menubar', 'none', ...
+	      'NumberTitle', 'off', ...
+	      'Visible', 'off');
+  image(X);
+  axis image;
+  ax = findobj(f, 'type', 'axes');
+  set(ax, 'Units', get(H,'PaperUnits'), ...
+	  'Position', [0 0 width height], ...
+	  'Visible', 'off');
+  set(ax, 'Units', 'pixels');
+  axesPos = get(ax,'Position');
+  figPos = get(f,'Position');
+  rootSize = get(0,'ScreenSize');
+  figPos(3:4) = axesPos(3:4);
+  if figPos(1) + figPos(3) > rootSize(3)
+    figPos(1) = rootSize(3) - figPos(3) - 50;
+  end
+  if figPos(2) + figPos(4) > rootSize(4)
+    figPos(2) = rootSize(4) - figPos(4) - 50;
+  end
+  set(f, 'Position',figPos, ...
+	 'Visible', 'on');
+end
+
+%
+%  Local Functions
+%
+
+function outData = LocalPushOldData(inData, objs, prop, values)
+outData.objs = {inData.objs{:}, objs};
+outData.prop = {inData.prop{:}, prop};
+outData.values = {inData.values{:}, values};
+
+function cellArray = LocalGetAsCell(fig,prop);
+cellArray = get(fig,prop);
+if (~isempty(cellArray)) & (~iscell(cellArray))
+  cellArray = {cellArray};
+end
+
+function newArray = LocalScale(inArray, scale, minValue)
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+  newArray{k} = max(minValue,scale*inArray{k}(1));
+end
+
+function newArray = LocalMapToGray(inArray);
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+  color = inArray{k};
+  if (~isempty(color))
+    if ischar(color)
+      switch color(1)
+       case 'y'
+	color = [1 1 0];
+       case 'm'
+	color = [1 0 1];
+       case 'c'
+	color = [0 1 1];
+       case 'r'
+	color = [1 0 0];
+       case 'g'
+	color = [0 1 0];
+       case 'b'
+	color = [0 0 1];
+       case 'w'
+	color = [1 1 1];
+       case 'k'
+	color = [0 0 0];
+       otherwise
+	newArray{k} = color;
+      end
+    end
+    if ~ischar(color)
+      color = 0.30*color(1) + 0.59*color(2) + 0.11*color(3);
+    end
+  end
+  if isempty(color) | ischar(color)
+    newArray{k} = color;
+  else
+    newArray{k} = [color color color];
+  end
+end
+
+function newArray = LocalMapCData(inArray);
+n = length(inArray);
+newArray = cell(n,1);
+for k=1:n
+  color = inArray{k};
+  if (ndims(color) == 3) & isa(color,'double')
+    gray = 0.30*color(:,:,1) + 0.59*color(:,:,2) + 0.11*color(:,:,3);
+    color(:,:,1) = gray;
+    color(:,:,2) = gray;
+    color(:,:,3) = gray;
+  end
+  newArray{k} = color;
+end
+
+function outData = LocalUpdateColors(inArray, prop, inData)
+value = LocalGetAsCell(inArray,prop);
+outData.objs = {inData.objs{:}, inArray};
+outData.prop = {inData.prop{:}, {prop}};
+outData.values = {inData.values{:}, value};
+if (~isempty(value))
+  if strcmp(prop,'CData') 
+    value = LocalMapCData(value);
+  else
+    value = LocalMapToGray(value);
+  end
+  set(inArray,{prop},value);
+end
+
+function bool = LocalIsPositiveScalar(value)
+bool = isnumeric(value) & ...
+       prod(size(value)) == 1 & ...
+       value > 0;
+
+function value = LocalToNum(value)
+if ischar(value)
+  value = str2num(value);
+end

modules/video_coding/codecs/test_framework/normal_async_test.cc

@@ -0,0 +1,563 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_async_test.h"
+#include "typedefs.h"
+#include <sstream>
+#include <assert.h>
+#include <queue>
+#include <string.h>
+#include "tick_util.h"
+
+using namespace webrtc;
+
+NormalAsyncTest::NormalAsyncTest()
+:
+_requestKeyFrame(false),
+_testNo(1),
+_appendNext(false),
+_decFrameCnt(0),
+_encFrameCnt(0),
+_missingFrames(false),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedPLI(false),
+_waitForKey(false),
+NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
+           _testNo)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(WebRtc_UWord32 bitRate)
+:
+_requestKeyFrame(false),
+_testNo(1),
+_appendNext(false),
+_decFrameCnt(0),
+_encFrameCnt(0),
+_missingFrames(false),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedPLI(false),
+_waitForKey(false),
+NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
+           bitRate, _testNo)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+                                 unsigned int testNo)
+:
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_decFrameCnt(0),
+_encFrameCnt(0),
+_missingFrames(false),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedPLI(false),
+_waitForKey(false),
+NormalTest(name, description, _testNo)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+                                 WebRtc_UWord32 bitRate, unsigned int testNo)
+:
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_decFrameCnt(0),
+_encFrameCnt(0),
+_missingFrames(false),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_rttFrames(0),
+_hasReceivedSLI(false),
+_hasReceivedPLI(false),
+_waitForKey(false),
+NormalTest(name, description, bitRate, _testNo)
+{
+}
+
+NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
+                                 WebRtc_UWord32 bitRate, unsigned int testNo,
+                                 unsigned int rttFrames)
+:
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+_decFrameCnt(0),
+_encFrameCnt(0),
+_missingFrames(false),
+_decodeCompleteTime(0),
+_encodeCompleteTime(0),
+_rttFrames(rttFrames),
+_hasReceivedSLI(false),
+_hasReceivedPLI(false),
+_waitForKey(false),
+NormalTest(name, description, bitRate, _testNo)
+{
+}
+
+void
+NormalAsyncTest::Setup()
+{
+    Test::Setup();
+    std::stringstream ss;
+    std::string strTestNo;
+    ss << _testNo;
+    ss >> strTestNo;
+
+    // Check if settings exist. Otherwise use defaults.
+    if (_outname == "")
+    {
+        _outname = "../../out_normaltest" + strTestNo + ".yuv";
+    }
+
+    if (_encodedName == "")
+    {
+        _encodedName = "../../encoded_normaltest" + strTestNo + ".yuv";
+    }
+
+    if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+    {
+        printf("Cannot read file %s.\n", _inname.c_str());
+        exit(1);
+    }
+
+    if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+    {
+        printf("Cannot write encoded file.\n");
+        exit(1);
+    }
+
+    char mode[3] = "wb";
+    if (_appendNext)
+    {
+        strncpy(mode, "ab", 3);
+    }
+
+    if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
+    {
+        printf("Cannot write file %s.\n", _outname.c_str());
+        exit(1);
+    }
+
+    _appendNext = true;
+}
+
+void
+NormalAsyncTest::Teardown()
+{
+    Test::Teardown();
+    fclose(_sourceFile);
+    fclose(_decodedFile);
+}
+
+FrameQueueTuple::~FrameQueueTuple()
+{
+    if (_codecSpecificInfo != NULL)
+    {
+        // TODO(holmer): implement virtual function for deleting this and
+        // remove warnings
+        delete _codecSpecificInfo;
+    }
+    if (_frame != NULL)
+    {
+        delete _frame;
+    }
+}
+
+void FrameQueue::PushFrame(TestVideoEncodedBuffer *frame,
+                           void* codecSpecificInfo)
+{
+    WriteLockScoped cs(_queueRWLock);
+    _frameBufferQueue.push(new FrameQueueTuple(frame, codecSpecificInfo));
+}
+
+FrameQueueTuple* FrameQueue::PopFrame()
+{
+    WriteLockScoped cs(_queueRWLock);
+    if (_frameBufferQueue.empty())
+    {
+        return NULL;
+    }
+    FrameQueueTuple* tuple = _frameBufferQueue.front();
+    _frameBufferQueue.pop();
+    return tuple;
+}
+
+bool FrameQueue::Empty()
+{
+    ReadLockScoped cs(_queueRWLock);
+    return _frameBufferQueue.empty();
+}
+
+WebRtc_UWord32 VideoEncodeCompleteCallback::EncodedBytes()
+{
+    return _encodedBytes;
+}
+
+WebRtc_Word32
+VideoEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
+                                     const void* codecSpecificInfo,
+                                     const webrtc::RTPFragmentationHeader*
+                                     fragmentation)
+{
+    _test.Encoded(encodedImage);
+    TestVideoEncodedBuffer *newBuffer = new TestVideoEncodedBuffer();
+    //newBuffer->VerifyAndAllocate(encodedImage._length);
+    newBuffer->VerifyAndAllocate(encodedImage._size);
+    _encodedBytes += encodedImage._length;
+    // If _frameQueue would have been a fixed sized buffer we could have asked
+    // it for an empty frame and then just do:
+    // emptyFrame->SwapBuffers(encodedBuffer);
+    // This is how it should be done in Video Engine to save in on memcpys
+    void* codecSpecificInfoCopy =
+        _test.CopyCodecSpecificInfo(codecSpecificInfo);
+    _test.CopyEncodedImage(*newBuffer, encodedImage, codecSpecificInfoCopy);
+    if (_encodedFile != NULL)
+    {
+        fwrite(newBuffer->GetBuffer(), 1, newBuffer->GetLength(), _encodedFile);
+    }
+    _frameQueue->PushFrame(newBuffer, codecSpecificInfoCopy);
+    return 0;
+}
+
+WebRtc_UWord32 VideoDecodeCompleteCallback::DecodedBytes()
+{
+    return _decodedBytes;
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::Decoded(RawImage& image)
+{
+    _test.Decoded(image);
+    _decodedBytes += image._length;
+    if (_decodedFile != NULL)
+    {
+        fwrite(image._buffer, 1, image._length, _decodedFile);
+    }
+    return 0;
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
+    const WebRtc_UWord64 pictureId)
+{
+    return _test.ReceivedDecodedReferenceFrame(pictureId);
+}
+
+WebRtc_Word32
+VideoDecodeCompleteCallback::ReceivedDecodedFrame(
+    const WebRtc_UWord64 pictureId)
+{
+    return _test.ReceivedDecodedFrame(pictureId);
+}
+
+void
+NormalAsyncTest::Encoded(const EncodedImage& encodedImage)
+{
+    _encodeCompleteTime = tGetTime();
+    _encFrameCnt++;
+    _totalEncodePipeTime += _encodeCompleteTime -
+        _encodeTimes[encodedImage._timeStamp];
+}
+
+void
+NormalAsyncTest::Decoded(const RawImage& decodedImage)
+{
+    _decodeCompleteTime = tGetTime();
+    _decFrameCnt++;
+    _totalDecodePipeTime += _decodeCompleteTime -
+        _decodeTimes[decodedImage._timeStamp];
+    _decodedWidth = decodedImage._width;
+    _decodedHeight = decodedImage._height;
+}
+
+void
+NormalAsyncTest::Perform()
+{
+    _inname = "test/testFiles/foreman_cif.yuv";
+    CodecSettings(352, 288, 30, _bitRate);
+    Setup();
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    if(_encoder->InitEncode(&_inst, 1, 1440) < 0)
+    {
+        exit(EXIT_FAILURE);
+    }
+    _decoder->InitDecode(&_inst, 1);
+    FrameQueue frameQueue;
+    VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+    VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
+    _encoder->RegisterEncodeCompleteCallback(&encCallback);
+    _decoder->RegisterDecodeCompleteCallback(&decCallback);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+    _totalEncodeTime = _totalDecodeTime = 0;
+    _totalEncodePipeTime = _totalDecodePipeTime = 0;
+    bool complete = false;
+    _framecnt = 0;
+    _encFrameCnt = 0;
+    _decFrameCnt = 0;
+    _sumEncBytes = 0;
+    _lengthEncFrame = 0;
+    double starttime = tGetTime();
+    while (!complete)
+    {
+        CodecSpecific_InitBitrate();
+        complete = Encode();
+        if (!frameQueue.Empty() || complete)
+        {
+            while (!frameQueue.Empty())
+            {
+                _frameToDecode =
+                    static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+                int lost = DoPacketLoss();
+                if (lost == 2)
+                {
+                    // Lost the whole frame, continue
+                    _missingFrames = true;
+                    delete _frameToDecode;
+                    _frameToDecode = NULL;
+                    continue;
+                }
+                int ret = Decode(lost);
+                delete _frameToDecode;
+                _frameToDecode = NULL;
+                if (ret < 0)
+                {
+                    fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+                    exit(EXIT_FAILURE);
+                }
+                else if (ret == 0)
+                {
+                    _framecnt++;
+                }
+                else
+                {
+                    fprintf(stderr,
+                        "\n\nPositive return value from decode!\n\n");
+                }
+            }
+        }
+    }
+    double endtime = tGetTime();
+    double totalExecutionTime = endtime - starttime;
+    printf("Total execution time: %.1f s\n", totalExecutionTime);
+    _sumEncBytes = encCallback.EncodedBytes();
+    double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
+    double avgEncTime = _totalEncodeTime / _encFrameCnt;
+    double avgDecTime = _totalDecodeTime / _decFrameCnt;
+    printf("Actual bitrate: %f kbps\n", actualBitRate);
+    printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
+    printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
+    printf("Average encode pipeline time: %.1f ms\n",
+           1000 * _totalEncodePipeTime / _encFrameCnt);
+    printf("Average decode pipeline  time: %.1f ms\n",
+           1000 * _totalDecodePipeTime / _decFrameCnt);
+    printf("Number of encoded frames: %u\n", _encFrameCnt);
+    printf("Number of decoded frames: %u\n", _decFrameCnt);
+    (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
+        _bitRate << " kbps" << std::endl;
+    (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+    (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+    _encoder->Release();
+    _decoder->Release();
+    Teardown();
+}
+
+bool
+NormalAsyncTest::Encode()
+{
+    _lengthEncFrame = 0;
+    fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+    _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+    _inputVideoBuffer.SetTimeStamp((unsigned int)
+        (_encFrameCnt * 9e4 / _inst.maxFramerate));
+    _inputVideoBuffer.SetWidth(_inst.width);
+    _inputVideoBuffer.SetHeight(_inst.height);
+    RawImage rawImage;
+    VideoBufferToRawImage(_inputVideoBuffer, rawImage);
+    if (feof(_sourceFile) != 0)
+    {
+        return true;
+    }
+    _encodeCompleteTime = 0;
+    _encodeTimes[rawImage._timeStamp] = tGetTime();
+    VideoFrameType frameType = kDeltaFrame;
+
+    // check SLI queue
+    _hasReceivedSLI = false;
+    while (!_signalSLI.empty() && _signalSLI.front().delay == 0)
+    {
+        // SLI message has arrived at sender side
+        _hasReceivedSLI = true;
+        _pictureIdSLI = _signalSLI.front().id;
+        _signalSLI.pop_front();
+    }
+    // decrement SLI queue times
+    for (std::list<fbSignal>::iterator it = _signalSLI.begin();
+        it !=_signalSLI.end(); it++)
+    {
+        (*it).delay--;
+    }
+
+    // check PLI queue
+    _hasReceivedPLI = false;
+    while (!_signalPLI.empty() && _signalPLI.front().delay == 0)
+    {
+        // PLI message has arrived at sender side
+        _hasReceivedPLI = true;
+        _signalPLI.pop_front();
+    }
+    // decrement PLI queue times
+    for (std::list<fbSignal>::iterator it = _signalPLI.begin();
+        it != _signalPLI.end(); it++)
+    {
+        (*it).delay--;
+    }
+
+    if (_hasReceivedPLI)
+    {
+        // respond to PLI by encoding a key frame
+        frameType = kKeyFrame;
+        _hasReceivedPLI = false;
+        _hasReceivedSLI = false; // don't trigger both at once
+    }
+
+    void* codecSpecificInfo = CreateEncoderSpecificInfo();
+    int ret = _encoder->Encode(rawImage, codecSpecificInfo, frameType);
+    if (codecSpecificInfo != NULL)
+    {
+        // TODO(holmer): implement virtual function for deleting this and
+        // remove warnings
+        delete codecSpecificInfo;
+        codecSpecificInfo = NULL;
+    }
+    if (_encodeCompleteTime > 0)
+    {
+        _totalEncodeTime += _encodeCompleteTime -
+            _encodeTimes[rawImage._timeStamp];
+    }
+    else
+    {
+        _totalEncodeTime += tGetTime() - _encodeTimes[rawImage._timeStamp];
+    }
+    assert(ret >= 0);
+    return false;
+}
+
+int
+NormalAsyncTest::Decode(int lossValue)
+{
+    _sumEncBytes += _frameToDecode->_frame->GetLength();
+    double starttime = 0;
+    EncodedImage encodedImage;
+    VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+    encodedImage._completeFrame = !lossValue;
+    _decodeCompleteTime = 0;
+    _decodeTimes[encodedImage._timeStamp] = tGetTime();
+    int ret = WEBRTC_VIDEO_CODEC_OK;
+    if (!_waitForKey || encodedImage._frameType == kKeyFrame)
+    {
+        _waitForKey = false;
+        ret = _decoder->Decode(encodedImage, _missingFrames,
+            _frameToDecode->_codecSpecificInfo);
+
+        if (ret >= 0)
+        {
+            _missingFrames = false;
+        }
+    }
+
+    // check for SLI
+    if (ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
+    {
+        // add an SLI feedback to the feedback "queue"
+        // to be delivered to encoder with _rttFrames delay
+        _signalSLI.push_back(fbSignal(_rttFrames,
+            static_cast<WebRtc_UWord8>((_lastDecPictureId) & 0x3f))); // 6 lsb
+
+        ret = WEBRTC_VIDEO_CODEC_OK;
+    }
+    else if (ret == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI)
+    {
+        // add an SLI feedback to the feedback "queue"
+        // to be delivered to encoder with _rttFrames delay
+        _signalSLI.push_back(fbSignal(_rttFrames,
+            static_cast<WebRtc_UWord8>((_lastDecPictureId + 1) & 0x3f)));//6 lsb
+
+        ret = WEBRTC_VIDEO_CODEC_OK;
+    }
+    else if (ret == WEBRTC_VIDEO_CODEC_ERROR)
+    {
+        // wait for new key frame
+        // add an PLI feedback to the feedback "queue"
+        // to be delivered to encoder with _rttFrames delay
+        _signalPLI.push_back(fbSignal(_rttFrames, 0 /* picId not used*/));
+        _waitForKey = true;
+
+        ret = WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    if (_decodeCompleteTime > 0)
+    {
+        _totalDecodeTime += _decodeCompleteTime -
+            _decodeTimes[encodedImage._timeStamp];
+    }
+    else
+    {
+        _totalDecodeTime += tGetTime() - _decodeTimes[encodedImage._timeStamp];
+    }
+    return ret;
+}
+
+void NormalAsyncTest::CodecSpecific_InitBitrate()
+{
+    if (_bitRate == 0)
+    {
+        _encoder->SetRates(600, _inst.maxFramerate);
+    }
+    else
+    {
+        _encoder->SetRates(_bitRate, _inst.maxFramerate);
+    }
+}
+
+void NormalAsyncTest::CopyEncodedImage(TestVideoEncodedBuffer& dest,
+                                       EncodedImage& src,
+                                       void* /*codecSpecificInfo*/) const
+{
+    dest.CopyBuffer(src._length, src._buffer);
+    dest.SetFrameType(src._frameType);
+    dest.SetCaptureWidth((WebRtc_UWord16)src._encodedWidth);
+    dest.SetCaptureHeight((WebRtc_UWord16)src._encodedHeight);
+    dest.SetTimeStamp(src._timeStamp);
+}
+double
+NormalAsyncTest::tGetTime()
+{// return time in sec
+    return ((double) (TickTime::MillisecondTimestamp())/1000);
+ }

modules/video_coding/codecs/test_framework/normal_async_test.h

@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
+
+#include "common_types.h"
+
+#include "normal_test.h"
+#include "rw_lock_wrapper.h"
+#include <list>
+#include <map>
+#include <queue>
+
+class FrameQueueTuple
+{
+public:
+    FrameQueueTuple(TestVideoEncodedBuffer *frame,
+                    const void* codecSpecificInfo = NULL)
+    :
+        _frame(frame),
+        _codecSpecificInfo(codecSpecificInfo)
+    {};
+    ~FrameQueueTuple();
+    TestVideoEncodedBuffer*   _frame;
+    const void*               _codecSpecificInfo;
+};
+
+class FrameQueue
+{
+public:
+    FrameQueue()
+    :
+        _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()),
+        _prevTS(-1)
+    {
+    }
+
+    ~FrameQueue()
+    {
+        delete &_queueRWLock;
+    }
+
+    void PushFrame(TestVideoEncodedBuffer *frame,
+                   void* codecSpecificInfo = NULL);
+    FrameQueueTuple* PopFrame();
+    bool Empty();
+
+private:
+    webrtc::RWLockWrapper&                       _queueRWLock;
+    std::queue<FrameQueueTuple *>     _frameBufferQueue;
+    WebRtc_Word64                       _prevTS;
+};
+
+// feedback signal to encoder
+struct fbSignal
+{
+    fbSignal(int d, WebRtc_UWord8 pid) : delay(d), id(pid) {};
+    int         delay;
+    WebRtc_UWord8 id;
+};
+
+class NormalAsyncTest : public NormalTest
+{
+public:
+    NormalAsyncTest();
+    NormalAsyncTest(WebRtc_UWord32 bitRate);
+    NormalAsyncTest(std::string name, std::string description,
+                    unsigned int testNo);
+    NormalAsyncTest(std::string name, std::string description,
+                    WebRtc_UWord32 bitRate, unsigned int testNo);
+    NormalAsyncTest(std::string name, std::string description,
+                    WebRtc_UWord32 bitRate, unsigned int testNo,
+                    unsigned int rttFrames);
+    virtual ~NormalAsyncTest() {};
+    virtual void Perform();
+    virtual void Encoded(const webrtc::EncodedImage& encodedImage);
+    virtual void Decoded(const webrtc::RawImage& decodedImage);
+    virtual void*
+    CopyCodecSpecificInfo(const void* /*codecSpecificInfo */) const
+    { return NULL; };
+    virtual void CopyEncodedImage(TestVideoEncodedBuffer& dest,
+                                  webrtc::EncodedImage& src,
+                                  void* /*codecSpecificInfo*/) const;
+    virtual void* CreateEncoderSpecificInfo() const { return NULL; };
+    virtual WebRtc_Word32
+    ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) { return 0;};
+    virtual WebRtc_Word32
+    ReceivedDecodedFrame(const WebRtc_UWord64 pictureId) { return 0;};
+
+protected:
+    virtual void Setup();
+    virtual void Teardown();
+    virtual bool Encode();
+    virtual int Decode(int lossValue = 0);
+    virtual void CodecSpecific_InitBitrate();
+    virtual int SetCodecSpecificParameters() {return 0;};
+    double tGetTime();// return time in sec
+
+    FILE*                   _sourceFile;
+    FILE*                   _decodedFile;
+    WebRtc_UWord32          _decodedWidth;
+    WebRtc_UWord32          _decodedHeight;
+    double                  _totalEncodeTime;
+    double                  _totalDecodeTime;
+    double                  _decodeCompleteTime;
+    double                  _encodeCompleteTime;
+    double                  _totalEncodePipeTime;
+    double                  _totalDecodePipeTime;
+    int                     _framecnt;
+    int                     _encFrameCnt;
+    int                     _decFrameCnt;
+    bool                    _requestKeyFrame;
+    unsigned int            _testNo;
+    unsigned int            _lengthEncFrame;
+    FrameQueueTuple*        _frameToDecode;
+    bool                    _appendNext;
+    std::map<WebRtc_UWord32, double> _encodeTimes;
+    std::map<WebRtc_UWord32, double> _decodeTimes;
+    bool                    _missingFrames;
+    std::list<fbSignal>     _signalSLI;
+    int                     _rttFrames;
+    mutable bool            _hasReceivedSLI;
+    WebRtc_UWord8           _pictureIdSLI;
+    WebRtc_UWord64          _lastDecPictureId;
+    std::list<fbSignal>     _signalPLI;
+    bool                    _hasReceivedPLI;
+    bool                    _waitForKey;
+};
+
+class VideoEncodeCompleteCallback : public webrtc::EncodedImageCallback
+{
+public:
+    VideoEncodeCompleteCallback(FILE* encodedFile, FrameQueue *frameQueue,
+                                NormalAsyncTest& test)
+    :
+      _encodedFile(encodedFile),
+      _frameQueue(frameQueue),
+      _test(test),
+      _encodedBytes(0)
+    {}
+
+    WebRtc_Word32
+    Encoded(webrtc::EncodedImage& encodedImage,
+            const void* codecSpecificInfo = NULL,
+            const webrtc::RTPFragmentationHeader* fragmentation = NULL);
+    WebRtc_UWord32 EncodedBytes();
+private:
+    FILE*             _encodedFile;
+    FrameQueue*       _frameQueue;
+    NormalAsyncTest&  _test;
+    WebRtc_UWord32    _encodedBytes;
+};
+
+class VideoDecodeCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+    VideoDecodeCompleteCallback(FILE* decodedFile, NormalAsyncTest& test)
+    :
+        _decodedFile(decodedFile),
+        _test(test),
+        _decodedBytes(0)
+    {}
+
+    virtual WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
+    virtual WebRtc_Word32
+    ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+    virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
+
+    WebRtc_UWord32 DecodedBytes();
+private:
+    FILE* _decodedFile;
+    NormalAsyncTest& _test;
+    WebRtc_UWord32    _decodedBytes;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_

modules/video_coding/codecs/test_framework/normal_test.cc

@@ -0,0 +1,246 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_test.h"
+#include <time.h>
+#include <sstream>
+#include <string.h>
+
+NormalTest::NormalTest()
+:
+_testNo(1),
+_lengthEncFrame(0),
+_appendNext(false),
+Test("Normal Test 1", "A test of normal execution of the codec")
+{
+}
+
+NormalTest::NormalTest(std::string name, std::string description, unsigned int testNo)
+:
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+Test(name, description)
+{
+}
+
+NormalTest::NormalTest(std::string name, std::string description, WebRtc_UWord32 bitRate, unsigned int testNo)
+:
+_requestKeyFrame(false),
+_testNo(testNo),
+_lengthEncFrame(0),
+_appendNext(false),
+Test(name, description, bitRate)
+{
+}
+
+void
+NormalTest::Setup()
+{
+    Test::Setup();
+    std::stringstream ss;
+    std::string strTestNo;
+    ss << _testNo;
+    ss >> strTestNo;
+
+    // Check if settings exist. Otherwise use defaults.
+    if (_outname == "")
+    {
+        _outname = "../../out_normaltest" + strTestNo + ".yuv";
+    }
+
+    if (_encodedName == "")
+    {
+        _encodedName = "../../encoded_normaltest" + strTestNo + ".yuv";
+    }
+    
+    if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
+    {
+        printf("Cannot read file %s.\n", _inname.c_str());
+        exit(1);
+    }
+
+    if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
+    {
+        printf("Cannot write encoded file.\n");
+        exit(1);
+    }
+
+    char mode[3] = "wb";
+    if (_appendNext)
+    {
+        strncpy(mode, "ab", 3);
+    }
+
+    if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
+    {
+        printf("Cannot write file %s.\n", _outname.c_str());
+        exit(1);
+    }
+
+    _appendNext = true;
+}
+
+void
+NormalTest::Teardown()
+{
+    Test::Teardown();
+    fclose(_sourceFile);
+    fclose(_decodedFile);
+}
+
+void
+NormalTest::Perform()
+{
+    _inname = "../../../../testFiles/foreman.yuv";
+    CodecSettings(352, 288, 30, _bitRate);
+    Setup();
+
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    
+    _encoder->InitEncode(&_inst, 1, 1460);
+    CodecSpecific_InitBitrate();
+    _decoder->InitDecode(&_inst,1);
+
+    _totalEncodeTime = _totalDecodeTime = 0;
+    _framecnt = 0;
+    _sumEncBytes = 0;
+    _lengthEncFrame = 0;
+    int decodeLength = 0;
+    while (!Encode())
+    {
+        DoPacketLoss();
+        _encodedVideoBuffer.UpdateLength(_encodedVideoBuffer.GetLength());
+        fwrite(_encodedVideoBuffer.GetBuffer(), 1, _encodedVideoBuffer.GetLength(), _encodedFile);
+        decodeLength = Decode();
+        if (decodeLength < 0)
+        {
+            fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
+            exit(EXIT_FAILURE);
+        }
+        fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
+        CodecSpecific_InitBitrate();
+        _framecnt++;
+    }
+
+    // Ensure we empty the decoding queue.
+    while (decodeLength > 0)
+    {
+        decodeLength = Decode();
+        if (decodeLength < 0)
+        {
+            fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
+            exit(EXIT_FAILURE);
+        }
+        fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
+    }
+
+    double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
+    double avgEncTime = _totalEncodeTime / _framecnt;
+    double avgDecTime = _totalDecodeTime / _framecnt;
+    printf("Actual bitrate: %f kbps\n", actualBitRate);
+    printf("Average encode time: %f s\n", avgEncTime);
+    printf("Average decode time: %f s\n", avgDecTime);
+    (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
+    (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+    (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+
+    _inputVideoBuffer.Free();
+    _encodedVideoBuffer.Reset();
+    _decodedVideoBuffer.Free();
+
+    _encoder->Release();
+    _decoder->Release();
+
+    Teardown();
+}
+
+bool
+NormalTest::Encode()
+{
+    _lengthEncFrame = 0;
+    fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+    if (feof(_sourceFile) != 0)
+    {
+        return true;
+    }
+    _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+    _inputVideoBuffer.SetTimeStamp(_framecnt);
+
+    // This multiple attempt ridiculousness is to accomodate VP7:
+    // 1. The wrapper can unilaterally reduce the framerate for low bitrates.
+    // 2. The codec inexplicably likes to reject some frames. Perhaps there
+    //    is a good reason for this...
+    int encodingAttempts = 0;
+    double starttime = 0;
+    double endtime = 0;
+    while (_lengthEncFrame == 0)
+    {
+        starttime = clock()/(double)CLOCKS_PER_SEC;
+
+        _inputVideoBuffer.SetWidth(_inst.width);
+        _inputVideoBuffer.SetHeight(_inst.height);
+        //_lengthEncFrame = _encoder->Encode(_inputVideoBuffer, _encodedVideoBuffer, _frameInfo,
+        //  _inst.frameRate, _requestKeyFrame && !(_framecnt%50));
+
+        endtime = clock()/(double)CLOCKS_PER_SEC;
+
+        _encodedVideoBuffer.SetCaptureHeight(_inst.height);
+        _encodedVideoBuffer.SetCaptureWidth(_inst.width);
+        if (_lengthEncFrame < 0)
+        {
+            (*_log) << "Error in encoder: " << _lengthEncFrame << std::endl;
+            fprintf(stderr,"\n\nError in encoder: %d\n\n", _lengthEncFrame);
+            exit(EXIT_FAILURE);
+        }
+        _sumEncBytes += _lengthEncFrame;
+
+        encodingAttempts++;
+        if (encodingAttempts > 50)
+        {
+            (*_log) << "Unable to encode frame: " << _framecnt << std::endl;
+            fprintf(stderr,"\n\nUnable to encode frame: %d\n\n", _framecnt);
+            exit(EXIT_FAILURE);
+        }
+    }
+    _totalEncodeTime += endtime - starttime;
+
+    if (encodingAttempts > 1)
+    {
+        (*_log) << encodingAttempts << " attempts required to encode frame: " <<
+            _framecnt + 1 << std::endl;
+        fprintf(stderr,"\n%d attempts required to encode frame: %d\n", encodingAttempts,
+            _framecnt + 1);
+    }
+        
+    return false;
+}
+
+int
+NormalTest::Decode()
+{
+    double starttime = clock()/(double)CLOCKS_PER_SEC;
+    _encodedVideoBuffer.SetWidth(_inst.width);
+    _encodedVideoBuffer.SetHeight(_inst.height);
+    int lengthDecFrame = 0;
+    //int lengthDecFrame = _decoder->Decode(_encodedVideoBuffer, _decodedVideoBuffer);
+    //_totalDecodeTime += (double)((clock()/(double)CLOCKS_PER_SEC) - starttime);
+    if (lengthDecFrame < 0)
+    {
+        return lengthDecFrame;
+    }
+    _encodedVideoBuffer.Reset();
+    _encodedVideoBuffer.UpdateLength(0);
+    return lengthDecFrame;
+}
+

modules/video_coding/codecs/test_framework/normal_test.h

@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+
+#include "test.h"
+
+class NormalTest : public Test
+{
+public:
+    NormalTest();
+    NormalTest(std::string name, std::string description, unsigned int testNo);
+    NormalTest(std::string name, std::string description, WebRtc_UWord32 bitRate, unsigned int testNo);
+    virtual ~NormalTest() {};
+    virtual void Perform();
+
+protected:
+    virtual void Setup();
+    virtual void Teardown();
+    virtual bool Encode();
+    virtual int Decode();
+    virtual void CodecSpecific_InitBitrate()=0;
+    virtual int DoPacketLoss() {return 0;};
+
+    FILE*                   _sourceFile;
+    FILE*                   _decodedFile;
+    FILE*                   _encodedFile;
+    double                  _totalEncodeTime;
+    double                  _totalDecodeTime;
+    unsigned int            _framecnt;
+    bool                    _requestKeyFrame;
+    unsigned int            _testNo;
+    int                     _lengthEncFrame;
+    bool                    _appendNext;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
+

modules/video_coding/codecs/test_framework/packet_loss_test.cc

@@ -0,0 +1,244 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet_loss_test.h"
+#include "video_source.h"
+#include <sstream>
+#include <cassert>
+#include <string.h>
+
+using namespace webrtc;
+
+PacketLossTest::PacketLossTest()
+:
+_lossRate(0.1),
+_lossProbability(0.1),
+_lastFrame(NULL),
+_lastFrameLength(0),
+NormalAsyncTest("PacketLossTest", "Encode, remove lost packets, decode", 300, 5)
+{
+}
+
+PacketLossTest::PacketLossTest(std::string name, std::string description)
+:
+_lossRate(0.1),
+_lossProbability(0.1),
+_lastFrame(NULL),
+_lastFrameLength(0),
+NormalAsyncTest(name, description, 300, 5)
+{
+}
+
+PacketLossTest::PacketLossTest(std::string name, std::string description, double lossRate, bool useNack, unsigned int rttFrames /* = 0*/)
+:
+_lossRate(lossRate),
+_lastFrame(NULL),
+_lastFrameLength(0),
+NormalAsyncTest(name, description, 300, 5, rttFrames)
+{
+    assert(lossRate >= 0 && lossRate <= 1);
+    if (useNack)
+    {
+        _lossProbability = 0;
+    }
+    else
+    {
+        _lossProbability = lossRate;
+    }
+}
+
+void
+PacketLossTest::Encoded(const EncodedImage& encodedImage)
+{
+    // push timestamp to queue
+    _frameQueue.push_back(encodedImage._timeStamp);
+    NormalAsyncTest::Encoded(encodedImage);
+}
+
+void
+PacketLossTest::Decoded(const RawImage& decodedImage)
+{
+    // check the frame queue if any frames have gone missing
+    assert(!_frameQueue.empty()); // decoded frame is not in the queue
+    while(_frameQueue.front() < decodedImage._timeStamp)
+    {
+        // this frame is missing
+        // write previous decoded frame again (frame freeze)
+        if (_decodedFile && _lastFrame)
+        {
+            fwrite(_lastFrame, 1, _lastFrameLength, _decodedFile);
+        }
+
+        // remove frame from queue
+        _frameQueue.pop_front();
+    }
+    assert(_frameQueue.front() == decodedImage._timeStamp); // decoded frame is not in the queue
+
+    // pop the current frame
+    _frameQueue.pop_front();
+
+    // save image for future freeze-frame
+    if (_lastFrameLength < decodedImage._length)
+    {
+        if (_lastFrame) delete [] _lastFrame;
+
+        _lastFrame = new WebRtc_UWord8[decodedImage._length];
+    }
+    memcpy(_lastFrame, decodedImage._buffer, decodedImage._length);
+    _lastFrameLength = decodedImage._length;
+
+    NormalAsyncTest::Decoded(decodedImage);
+}
+
+void
+PacketLossTest::Teardown()
+{
+    if (_totalKept + _totalThrown > 0)
+    {
+        printf("Target packet loss rate: %.4f\n", _lossProbability);
+        printf("Actual packet loss rate: %.4f\n", (_totalThrown * 1.0f) / (_totalKept + _totalThrown));
+        printf("Channel rate: %.2f kbps\n",
+            0.001 * 8.0 * _sumChannelBytes / ((_framecnt * 1.0f) / _inst.maxFramerate));
+    }
+    else
+    {
+        printf("No packet losses inflicted\n");
+    }
+
+    NormalAsyncTest::Teardown();
+}
+
+void
+PacketLossTest::Setup()
+{
+    const VideoSource source(_inname, _inst.width, _inst.height, _inst.maxFramerate);
+
+    std::stringstream ss;
+    std::string lossRateStr;
+    ss << _lossRate;
+    ss >> lossRateStr;
+    _encodedName = "../../" + source.GetName() + "-" + lossRateStr;
+    _outname = "../../out-" + source.GetName() + "-" + lossRateStr;
+
+    if (_lossProbability != _lossRate)
+    {
+        _encodedName += "-nack";
+        _outname += "-nack";
+    }
+    _encodedName += ".vp8";
+    _outname += ".yuv";
+
+    _totalKept = 0;
+    _totalThrown = 0;
+    _sumChannelBytes = 0;
+
+    NormalAsyncTest::Setup();
+}
+
+void
+PacketLossTest::CodecSpecific_InitBitrate()
+{
+    assert(_bitRate > 0);
+    WebRtc_UWord32 simulatedBitRate;
+    if (_lossProbability != _lossRate)
+    {
+        // Simulating NACK
+        simulatedBitRate = WebRtc_UWord32(_bitRate / (1 + _lossRate));
+    }
+    else
+    {
+        simulatedBitRate = _bitRate;
+    }
+    _encoder->SetPacketLoss((WebRtc_UWord32)(_lossProbability * 255.0));
+    _encoder->SetRates(simulatedBitRate, _inst.maxFramerate);
+}
+
+int PacketLossTest::DoPacketLoss()
+{
+    // Only packet loss for delta frames
+    if (_frameToDecode->_frame->GetLength() == 0 || _frameToDecode->_frame->GetFrameType() != kDeltaFrame)
+    {
+        _sumChannelBytes += _frameToDecode->_frame->GetLength();
+        return 0;
+    }
+    //printf("Encoded: %d bytes\n", _encodedVideoBuffer.GetLength());
+    unsigned char *packet = NULL;
+    TestVideoEncodedBuffer newEncBuf;
+    newEncBuf.VerifyAndAllocate(_lengthSourceFrame);
+    _inBufIdx = 0;
+    _outBufIdx = 0;
+    int size = 1;
+    int kept = 0;
+    int thrown = 0;
+    int count = 0;
+    while ((size = NextPacket(1500, &packet)) > 0)
+    {
+        if (!PacketLoss(_lossProbability))
+        {
+            InsertPacket(&newEncBuf, packet, size);
+            kept++;
+        }
+        else
+        {
+            // Use the ByteLoss function if you want to lose only
+            // parts of a packet, and not the whole packet.
+
+            //int size2 = ByteLoss(size, packet, 15);
+            thrown++;
+            //if (size2 != size)
+            //{
+            //    InsertPacket(&newEncBuf, packet, size2);
+            //}
+        }
+    }
+    int	lossResult  = (thrown!=0);	// 0 = no loss	1 = loss(es)
+    if (lossResult)
+    {
+        lossResult += (kept==0);	// 2 = all lost = full frame
+    }
+    _frameToDecode->_frame->CopyBuffer(newEncBuf.GetLength(), newEncBuf.GetBuffer());
+    _sumChannelBytes += newEncBuf.GetLength();
+    _totalKept += kept;
+    _totalThrown += thrown;
+    return lossResult;
+    //printf("Threw away: %d out of %d packets\n", thrown, thrown + kept);
+    //printf("Encoded left: %d bytes\n", _encodedVideoBuffer.GetLength());
+}
+
+int PacketLossTest::NextPacket(int mtu, unsigned char **pkg)
+{
+    unsigned char *buf = _frameToDecode->_frame->GetBuffer();
+    *pkg = buf + _inBufIdx;
+    if (static_cast<long>(_frameToDecode->_frame->GetLength()) - _inBufIdx <= mtu)
+    {
+        int size = _frameToDecode->_frame->GetLength() - _inBufIdx;
+        _inBufIdx = _frameToDecode->_frame->GetLength();
+        return size;
+    }
+    _inBufIdx += mtu;
+    return mtu;
+}
+
+int PacketLossTest::ByteLoss(int size, unsigned char *pkg, int bytesToLose)
+{
+    return size;
+}
+
+void PacketLossTest::InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size)
+{
+    if (static_cast<long>(buf->GetSize()) - _outBufIdx < size)
+    {
+        printf("InsertPacket error!\n");
+        return;
+    }
+    memcpy(buf->GetBuffer() + _outBufIdx, pkg, size);
+    buf->UpdateLength(buf->GetLength() + size);
+    _outBufIdx += size;
+}

modules/video_coding/codecs/test_framework/packet_loss_test.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
+
+#include <list>
+
+#include "normal_async_test.h"
+
+class PacketLossTest : public NormalAsyncTest
+{
+public:
+    PacketLossTest();
+    virtual ~PacketLossTest() {if(_lastFrame) {delete [] _lastFrame; _lastFrame = NULL;}}
+    virtual void Encoded(const webrtc::EncodedImage& encodedImage);
+    virtual void Decoded(const webrtc::RawImage& decodedImage);
+protected:
+    PacketLossTest(std::string name, std::string description);
+    PacketLossTest(std::string name,
+                   std::string description,
+                   double lossRate,
+                   bool useNack,
+                   unsigned int rttFrames = 0);
+
+    virtual void Setup();
+    virtual void Teardown();
+    virtual void CodecSpecific_InitBitrate();
+    virtual int DoPacketLoss();
+    virtual int NextPacket(int size, unsigned char **pkg);
+    virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
+    virtual void InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size);
+    int _inBufIdx;
+    int _outBufIdx;
+
+    // When NACK is being simulated _lossProbabilty is zero,
+    // otherwise it is set equal to _lossRate.
+    // Desired channel loss rate.
+    double _lossRate;
+    // Probability used to simulate packet drops.
+    double _lossProbability;
+
+    int _totalKept;
+    int _totalThrown;
+    int _sumChannelBytes;
+    std::list<WebRtc_UWord32> _frameQueue;
+    WebRtc_UWord8* _lastFrame;
+    WebRtc_UWord32 _lastFrameLength;
+};
+
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_

modules/video_coding/codecs/test_framework/performance_test.cc

@@ -0,0 +1,290 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "performance_test.h"
+#include "tick_util.h"
+#include <assert.h>
+
+using namespace webrtc;
+
+#define NUM_FRAMES 300
+
+PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate)
+:
+_numCodecs(0),
+_tests(NULL),
+_encoders(NULL),
+_decoders(NULL),
+_threads(NULL),
+_rawImageLock(NULL),
+_encodeEvents(new EventWrapper*[1]),
+_stopped(true),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL),
+NormalAsyncTest(bitRate)
+{
+}
+
+PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs)
+:
+_numCodecs(numCodecs),
+_tests(new PerformanceTest*[_numCodecs]),
+_encoders(new VideoEncoder*[_numCodecs]),
+_decoders(new VideoDecoder*[_numCodecs]),
+_threads(new ThreadWrapper*[_numCodecs]),
+_rawImageLock(RWLockWrapper::CreateRWLock()),
+_encodeEvents(new EventWrapper*[_numCodecs]),
+_stopped(true),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL),
+NormalAsyncTest(bitRate)
+{
+    for (int i=0; i < _numCodecs; i++)
+    {
+        _tests[i] = new PerformanceTest(bitRate);
+        _encodeEvents[i] = EventWrapper::Create();
+    }
+}
+
+PerformanceTest::~PerformanceTest()
+{
+    if (_encoders != NULL)
+    {
+        delete [] _encoders;
+    }
+    if (_decoders != NULL)
+    {
+        delete [] _decoders;
+    }
+    if (_tests != NULL)
+    {
+        delete [] _tests;
+    }
+    if (_threads != NULL)
+    {
+        delete [] _threads;
+    }
+    if (_rawImageLock != NULL)
+    {
+        delete _rawImageLock;
+    }
+    if (_encodeEvents != NULL)
+    {
+        delete [] _encodeEvents;
+    }
+}
+
+void
+PerformanceTest::Setup()
+{
+    _inname = "../../../../testFiles/foreman.yuv";
+    NormalAsyncTest::Setup(); // Setup input and output files
+    CodecSettings(352, 288, 30, _bitRate); // common to all codecs
+    for (int i=0; i < _numCodecs; i++)
+    {
+        _encoders[i] = CreateEncoder();
+        _decoders[i] = CreateDecoder();
+        if (_encoders[i] == NULL)
+        {
+            printf("Must create a codec specific test!\n");
+            exit(EXIT_FAILURE);
+        }
+        if(_encoders[i]->InitEncode(&_inst, 4, 1440) < 0)
+        {
+            exit(EXIT_FAILURE);
+        }
+        if (_decoders[i]->InitDecode(&_inst, 1))
+        {
+            exit(EXIT_FAILURE);
+        }
+        _tests[i]->SetEncoder(_encoders[i]);
+        _tests[i]->SetDecoder(_decoders[i]);
+        _tests[i]->_rawImageLock = _rawImageLock;
+        _encodeEvents[i]->Reset();
+        _tests[i]->_encodeEvents[0] = _encodeEvents[i];
+        _tests[i]->_inst = _inst;
+        _threads[i] = ThreadWrapper::CreateThread(PerformanceTest::RunThread, _tests[i]);
+        unsigned int id = 0;
+        _tests[i]->_stopped = false;
+        _threads[i]->Start(id);
+    }
+}
+
+void
+PerformanceTest::Perform()
+{
+    Setup();
+    EventWrapper& sleepEvent = *EventWrapper::Create();
+    const WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+    for (int i=0; i < NUM_FRAMES; i++)
+    {
+        {
+            // Read a new frame from file
+            WriteLockScoped imageLock(*_rawImageLock);
+            _lengthEncFrame = 0;
+            fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+            if (feof(_sourceFile) != 0)
+            {
+                rewind(_sourceFile);
+            }
+            _inputVideoBuffer.VerifyAndAllocate(_inst.width*_inst.height*3/2);
+            _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+            _inputVideoBuffer.SetTimeStamp((unsigned int) (_encFrameCnt * 9e4 / static_cast<float>(_inst.maxFramerate)));
+            _inputVideoBuffer.SetWidth(_inst.width);
+            _inputVideoBuffer.SetHeight(_inst.height);
+            for (int i=0; i < _numCodecs; i++)
+            {
+                _tests[i]->_inputVideoBuffer.CopyPointer(_inputVideoBuffer);
+                _encodeEvents[i]->Set();
+            }
+        }
+        if (i < NUM_FRAMES - 1)
+        {
+            sleepEvent.Wait(33);
+        }
+    }
+    for (int i=0; i < _numCodecs; i++)
+    {
+        _tests[i]->_stopped = true;
+        _encodeEvents[i]->Set();
+        _threads[i]->Stop();
+    }
+    const WebRtc_UWord32 totalTime =
+            static_cast<WebRtc_UWord32>(TickTime::MillisecondTimestamp() - startTime);
+    printf("Total time: %u\n", totalTime);
+    delete &sleepEvent;
+    Teardown();
+}
+
+void PerformanceTest::Teardown()
+{
+    if (_encodeCompleteCallback != NULL)
+    {
+        delete _encodeCompleteCallback;
+    }
+    if (_decodeCompleteCallback != NULL)
+    {
+        delete _decodeCompleteCallback;
+    }
+    // main test only, all others have numCodecs = 0:
+    if (_numCodecs > 0)
+    {
+        WriteLockScoped imageLock(*_rawImageLock);
+        _inputVideoBuffer.Free();
+        NormalAsyncTest::Teardown();
+    }
+    for (int i=0; i < _numCodecs; i++)
+    {
+        _encoders[i]->Release();
+        delete _encoders[i];
+        _decoders[i]->Release();
+        delete _decoders[i];
+        _tests[i]->_inputVideoBuffer.ClearPointer();
+        _tests[i]->_rawImageLock = NULL;
+        _tests[i]->Teardown();
+        delete _tests[i];
+        delete _encodeEvents[i];
+        delete _threads[i];
+    }
+}
+
+bool
+PerformanceTest::RunThread(void* obj)
+{
+    PerformanceTest& test = *static_cast<PerformanceTest*>(obj);
+    return test.PerformSingleTest();
+}
+
+bool
+PerformanceTest::PerformSingleTest()
+{
+    if (_encodeCompleteCallback == NULL)
+    {
+        _encodeCompleteCallback = new VideoEncodeCompleteCallback(NULL, &_frameQueue, *this);
+        _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
+    }
+    if (_decodeCompleteCallback == NULL)
+    {
+        _decodeCompleteCallback = new VideoDecodeCompleteCallback(NULL, *this);
+        _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
+    }
+    (*_encodeEvents)->Wait(WEBRTC_EVENT_INFINITE); // The first event is used for every single test
+    CodecSpecific_InitBitrate();
+    bool complete = false;
+    {
+        ReadLockScoped imageLock(*_rawImageLock);
+        complete = Encode();
+    }
+    if (!_frameQueue.Empty() || complete)
+    {
+        while (!_frameQueue.Empty())
+        {
+            _frameToDecode = static_cast<FrameQueueTuple *>(_frameQueue.PopFrame());
+            int lost = DoPacketLoss();
+            if (lost == 2)
+            {
+                // Lost the whole frame, continue
+                _missingFrames = true;
+                delete _frameToDecode;
+                _frameToDecode = NULL;
+                continue;
+            }
+            int ret = Decode(lost);
+            delete _frameToDecode;
+            _frameToDecode = NULL;
+            if (ret < 0)
+            {
+                fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+                return false;
+            }
+            else if (ret < 0)
+            {
+                fprintf(stderr, "\n\nPositive return value from decode!\n\n");
+                return false;
+            }
+        }
+    }
+    if (_stopped)
+    {
+        return false;
+    }
+    return true;
+}
+
+bool PerformanceTest::Encode()
+{
+    RawImage rawImage;
+    VideoBufferToRawImage(_inputVideoBuffer, rawImage);
+    VideoFrameType frameType = kDeltaFrame;
+    if (_requestKeyFrame && !(_encFrameCnt%50))
+    {
+        frameType = kKeyFrame;
+    }
+    void* codecSpecificInfo = CreateEncoderSpecificInfo();
+    int ret = _encoder->Encode(rawImage, codecSpecificInfo, frameType);
+    if (codecSpecificInfo != NULL)
+    {
+        // TODO(holmer): implement virtual function for deleting this and remove warnings
+        delete codecSpecificInfo;
+        codecSpecificInfo = NULL;
+    }
+    assert(ret >= 0);
+    return false;
+}
+
+int PerformanceTest::Decode(int lossValue)
+{
+    EncodedImage encodedImage;
+    VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+    encodedImage._completeFrame = !lossValue;
+    int ret = _decoder->Decode(encodedImage, _missingFrames, _frameToDecode->_codecSpecificInfo);
+    _missingFrames = false;
+    return ret;
+}

modules/video_coding/codecs/test_framework/performance_test.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
+
+#include "normal_async_test.h"
+#include "thread_wrapper.h"
+#include "rw_lock_wrapper.h"
+#include "event_wrapper.h"
+
+class PerformanceTest : public NormalAsyncTest
+{
+public:
+    PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs);
+    virtual ~PerformanceTest();
+
+    virtual void Perform();
+    virtual void Print() {};
+
+protected:
+    PerformanceTest(WebRtc_UWord32 bitRate);
+    virtual void Setup();
+    virtual bool Encode();
+    virtual int Decode(int lossValue = 0);
+    virtual void Teardown();
+    static bool RunThread(void* obj);
+    bool PerformSingleTest();
+
+    virtual webrtc::VideoEncoder* CreateEncoder() const { return NULL; };
+    virtual webrtc::VideoDecoder* CreateDecoder() const { return NULL; };
+
+    WebRtc_UWord8                 _numCodecs;
+    PerformanceTest**             _tests;
+    webrtc::VideoEncoder**        _encoders;
+    webrtc::VideoDecoder**        _decoders;
+    webrtc::ThreadWrapper**       _threads;
+    webrtc::RWLockWrapper*        _rawImageLock;
+    webrtc::EventWrapper**        _encodeEvents;
+    FrameQueue                    _frameQueue;
+    bool                          _stopped;
+    webrtc::EncodedImageCallback* _encodeCompleteCallback;
+    webrtc::DecodedImageCallback* _decodeCompleteCallback;
+    FILE*                         _outFile;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_

modules/video_coding/codecs/test_framework/plotBenchmark.m

@@ -0,0 +1,427 @@
+function plotBenchmark(fileNames, export)
+%PLOTBENCHMARK Plots and exports video codec benchmarking results.
+%   PLOTBENCHMARK(FILENAMES, EXPORT) parses the video codec benchmarking result
+%   files given by the cell array of strings FILENAME. It plots the results and
+%   optionally exports each plot to an appropriately named file.
+%
+%   EXPORT parameter:
+%       'none'  No file exports.
+%       'eps'   Exports to eps files (default).
+%       'pdf'   Exports to eps files and uses the command-line utility
+%               epstopdf to obtain pdf files.
+%
+%   Example:
+%       plotBenchmark({'H264Benchmark.txt' 'LSVXBenchmark.txt'}, 'pdf')
+
+if (nargin < 1)
+    error('Too few input arguments');
+elseif (nargin < 2)
+    export = 'eps';
+end
+
+if ~iscell(fileNames)
+    if ischar(fileNames)
+        % one single file name as a string is ok
+        if size(fileNames,1) > 1
+            % this is a char matrix, not ok
+            error('First argument must not be a char matrix');
+        end
+        % wrap in a cell array
+        fileNames = {fileNames};
+    else
+        error('First argument must be a cell array of strings');
+    end
+end
+
+if ~ischar(export)
+    error('Second argument must be a string');
+end
+
+outpath = 'BenchmarkPlots';
+[status, errMsg] = mkdir(outpath);
+if status == 0
+    error(errMsg);
+end
+
+nCases = 0;
+testCases = [];
+% Read each test result file
+for fileIdx = 1:length(fileNames)
+    if ~isstr(fileNames{fileIdx})
+        error('First argument must be a cell array of strings');
+    end
+
+    fid = fopen(fileNames{fileIdx}, 'rt');
+    if fid == -1
+        error(['Unable to open ' fileNames{fileIdx}]);
+    end
+
+    version = '1.0';
+    if ~strcmp(fgetl(fid), ['#!benchmark' version])
+        fclose(fid);
+        error(['Requires benchmark file format version ' version]);
+    end
+
+    % Parse results file into testCases struct
+    codec = fgetl(fid);
+    tline = fgetl(fid);
+    while(tline ~= -1)
+        nCases = nCases + 1;
+
+        delim = strfind(tline, ',');
+        name = tline(1:delim(1)-1);
+        % Drop underscored suffix from name
+        underscore = strfind(name, '_'); 
+        if ~isempty(underscore)
+            name = name(1:underscore(1)-1);
+        end
+
+        resolution = tline(delim(1)+1:delim(2)-1);
+        frameRate = tline(delim(2)+1:end);
+
+        tline = fgetl(fid);
+        delim = strfind(tline, ',');
+        bitrateLabel = tline(1:delim(1)-1); 
+        bitrate = sscanf(tline(delim(1):end),',%f');
+
+        tline = fgetl(fid);
+        delim = strfind(tline, ',');
+        psnrLabel = tline(1:delim(1)-1); 
+        psnr = sscanf(tline(delim(1):end),',%f'); 
+
+
+        % Default data for the optional lines
+        speedLabel = 'Default';
+        speed = 0;
+        ssimLabel = 'Default';
+        ssim = 0;
+        
+        tline = fgetl(fid);
+        delim = strfind(tline, ',');
+        
+        while ~isempty(delim)
+            % More data
+            % Check type of data
+            if strncmp(lower(tline), 'speed', 5)
+                % Speed data included
+                speedLabel = tline(1:delim(1)-1);
+                speed = sscanf(tline(delim(1):end), ',%f');
+
+                tline = fgetl(fid);
+                
+            elseif strncmp(lower(tline), 'encode time', 11)
+                % Encode and decode times included
+                % TODO: take care of the data
+                
+                % pop two lines from file
+                tline = fgetl(fid);
+                tline = fgetl(fid);
+                
+            elseif strncmp(tline, 'SSIM', 4)
+                % SSIM data included
+                ssimLabel = tline(1:delim(1)-1);
+                ssim = sscanf(tline(delim(1):end), ',%f');
+
+                tline = fgetl(fid);
+            end
+            delim = strfind(tline, ',');
+        end
+
+        testCases = [testCases struct('codec', codec, 'name', name, 'resolution', ...
+            resolution, 'frameRate', frameRate, 'bitrate', bitrate, 'psnr', psnr, ...
+            'speed', speed, 'bitrateLabel', bitrateLabel, 'psnrLabel', psnrLabel, ...
+            'speedLabel', speedLabel, ...
+            'ssim', ssim, 'ssimLabel', ssimLabel)];
+
+        tline = fgetl(fid);
+    end
+
+    fclose(fid);
+end
+
+i = 0;
+casesPsnr = testCases;
+while ~isempty(casesPsnr)
+    i = i + 1;
+    casesPsnr = plotOnePsnr(casesPsnr, i, export, outpath);
+end
+
+casesSSIM = testCases;
+while ~isempty(casesSSIM)
+    i = i + 1;
+    casesSSIM = plotOneSSIM(casesSSIM, i, export, outpath);
+end
+
+casesSpeed = testCases;
+while ~isempty(casesSpeed)
+    if casesSpeed(1).speed == 0
+        casesSpeed = casesSpeed(2:end);
+    else
+        i = i + 1;
+        casesSpeed = plotOneSpeed(casesSpeed, i, export, outpath);
+    end
+end
+
+
+
+%%%%%%%%%%%%%%%%%%
+%% SUBFUNCTIONS %%
+%%%%%%%%%%%%%%%%%%
+
+function casesOut = plotOnePsnr(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+    if strcmp(cases(1).resolution, cases(i).resolution) & ...
+        strcmp(cases(1).frameRate, cases(i).frameRate)
+        plotIdx = [plotIdx i];
+    end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+    simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+    while ~isempty(simIndx)
+        diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
+        cases(i).psnr = cases(i).psnr(diffIndx);
+        cases(i).bitrate = cases(i).bitrate(diffIndx);
+        simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+    end
+end
+
+% Prepare figure with axis labels and so on
+hFig = figure(num);
+clf;
+hold on;
+grid on;
+axis([0 1100 20 50]);
+set(gca, 'XTick', 0:200:1000);
+set(gca, 'YTick', 20:10:60);
+xlabel(cases(1).bitrateLabel);
+ylabel(cases(1).psnrLabel);
+res = cases(1).resolution;
+frRate = cases(1).frameRate;
+title([res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+    i = i + 1;
+    [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'psnr', i, sequences, 1);
+
+    % Stored to generate the legend
+    hLines = [hLines ; hLine];
+    codecs = {codecs{:} codec};
+end
+legend(hLines, codecs, 4);
+hold off;
+
+if ~strcmp(export, 'none')
+    % Export figure to an eps file
+    res = stripws(res);
+    frRate = stripws(frRate);
+    exportName = [outpath '/psnr-' res '-' frRate];
+    exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+    % Use the epstopdf utility to convert to pdf
+    system(['epstopdf ' exportName '.eps']);  
+end
+
+
+function casesOut = plotOneSSIM(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+    if strcmp(cases(1).resolution, cases(i).resolution) & ...
+        strcmp(cases(1).frameRate, cases(i).frameRate)
+        plotIdx = [plotIdx i];
+    end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+    simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+    while ~isempty(simIndx)
+        diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
+        cases(i).ssim = cases(i).ssim(diffIndx);
+        cases(i).bitrate = cases(i).bitrate(diffIndx);
+        simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
+    end
+end
+
+% Prepare figure with axis labels and so on
+hFig = figure(num);
+clf;
+hold on;
+grid on;
+axis([0 1100 0.5 1]); % y-limit are set to 'auto' below
+set(gca, 'XTick', 0:200:1000);
+%set(gca, 'YTick', 20:10:60);
+xlabel(cases(1).bitrateLabel);
+ylabel(cases(1).ssimLabel);
+res = cases(1).resolution;
+frRate = cases(1).frameRate;
+title([res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+    i = i + 1;
+    [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'ssim', i, sequences, 1);
+
+    % Stored to generate the legend
+    hLines = [hLines ; hLine];
+    codecs = {codecs{:} codec};
+end
+%set(gca,'YLimMode','auto')
+set(gca,'YLim',[0.5 1])
+set(gca,'YScale','log')
+legend(hLines, codecs, 4);
+hold off;
+
+if ~strcmp(export, 'none')
+    % Export figure to an eps file
+    res = stripws(res);
+    frRate = stripws(frRate);
+    exportName = [outpath '/psnr-' res '-' frRate];
+    exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+    % Use the epstopdf utility to convert to pdf
+    system(['epstopdf ' exportName '.eps']);  
+end
+
+
+function casesOut = plotOneSpeed(cases, num, export, outpath)
+% Find matching specs
+plotIdx = 1;
+for i = 2:length(cases)
+    if strcmp(cases(1).resolution, cases(i).resolution) & ...
+        strcmp(cases(1).frameRate, cases(i).frameRate) & ...
+        strcmp(cases(1).name, cases(i).name)
+        plotIdx = [plotIdx i];
+    end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+% Prune similar results
+for i = 1:length(cases)
+    simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
+    while ~isempty(simIndx)
+        diffIndx = setdiff(1:length(cases(i).psnr), simIndx);
+        cases(i).psnr = cases(i).psnr(diffIndx);
+        cases(i).speed = cases(i).speed(diffIndx);
+        simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
+    end
+end
+
+hFig = figure(num);
+clf;
+hold on;
+%grid on;
+xlabel(cases(1).psnrLabel);
+ylabel(cases(1).speedLabel);
+res = cases(1).resolution;
+name = cases(1).name;
+frRate = cases(1).frameRate;
+title([name ', ' res ', ' frRate]);
+
+hLines = [];
+codecs = {};
+sequences = {};
+i = 0;
+while ~isempty(cases)
+    i = i + 1;
+    [cases, hLine, codec, sequences] = plotOneCodec(cases, 'psnr', 'speed', i, sequences, 0);
+
+    % Stored to generate the legend
+    hLines = [hLines ; hLine];
+    codecs = {codecs{:} codec};
+end
+legend(hLines, codecs, 1);
+hold off;
+
+if ~strcmp(export, 'none')
+    % Export figure to an eps file
+    res = stripws(res);
+    frRate = stripws(frRate);
+    exportName = [outpath '/speed-' name '-' res '-' frRate];
+    exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
+end
+
+if strcmp(export, 'pdf')
+    % Use the epstopdf utility to convert to pdf
+    system(['epstopdf ' exportName '.eps']);  
+end
+
+
+function [casesOut, hLine, codec, sequences] = plotOneCodec(cases, xfield, yfield, num, sequences, annotatePlot)
+plotStr = {'gx-', 'bo-', 'r^-', 'kd-', 'cx-', 'go--', 'b^--'};
+% Find matching codecs
+plotIdx = 1;
+for i = 2:length(cases)
+    if strcmp(cases(1).codec, cases(i).codec)
+        plotIdx = [plotIdx i];
+    end
+end
+
+% Return unplotted cases
+casesOut = cases(setdiff(1:length(cases), plotIdx));
+cases = cases(plotIdx);
+
+for i = 1:length(cases)
+    % Plot a single case
+    hLine = plot(getfield(cases(i), xfield), getfield(cases(i), yfield), plotStr{num}, ...
+        'LineWidth', 1.1, 'MarkerSize', 6);
+end
+
+% hLine handle and codec are returned to construct the legend afterwards
+codec = cases(1).codec;
+
+if annotatePlot == 0
+    return;
+end
+
+for i = 1:length(cases)
+    % Print the codec name as a text label
+    % Ensure each codec is only printed once
+    sequencePlotted = 0;
+    for j = 1:length(sequences)
+        if strcmp(cases(i).name, sequences{j})
+            sequencePlotted = 1;
+            break;
+        end
+    end
+
+    if sequencePlotted == 0
+        text(getfield(cases(i), xfield, {1}), getfield(cases(i), yfield, {1}), ...
+            ['    ' cases(i).name]);
+        sequences = {sequences{:} cases(i).name};
+    end
+end
+
+
+% Strip whitespace from string
+function str = stripws(str)
+if ~isstr(str)
+    error('String required');
+end
+str = str(setdiff(1:length(str), find(isspace(str) == 1)));

modules/video_coding/codecs/test_framework/test.cc

@@ -0,0 +1,534 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test.h"
+#include "video_source.h"
+#include "vplib.h"
+#include "event_wrapper.h"
+#include "thread_wrapper.h"
+#include <iostream>
+#include <fstream>
+#include <cmath>
+#include <ctime>
+#include <string.h>
+#include <cassert>
+#include <vector>
+
+using namespace webrtc;
+
+long filesize(const char *filename); // local function defined at end of file
+
+struct SSIMcontext
+{
+    SSIMcontext() :
+    refFileName(NULL), testFileName(NULL), width(0), height(0),
+        SSIMptr(NULL), startFrame(-1), endFrame(-1), evnt(NULL) {};
+    SSIMcontext(const char *ref, const char *test, int w, int h, double *Sptr,
+        int start, int end, EventWrapper* ev) :
+    refFileName(ref), testFileName(test), width(w), height(h),
+        SSIMptr(Sptr), startFrame(start), endFrame(end), evnt(ev) {};
+    const char *refFileName;
+    const char *testFileName;
+    int width;
+    int height;
+    double *SSIMptr;
+    int startFrame;
+    int endFrame;
+    EventWrapper* evnt;
+};
+
+Test::Test(std::string name, std::string description)
+:
+_name(name),
+_description(description),
+_bitRate(0),
+_inname(""),
+_outname(""),
+_encodedName("")
+{
+    memset(&_inst, 0, sizeof(_inst));
+    unsigned int seed = static_cast<unsigned int>(0);
+    std::srand(seed);
+}
+
+Test::Test(std::string name, std::string description, WebRtc_UWord32 bitRate)
+:
+_name(name),
+_description(description),
+_bitRate(bitRate),
+_inname(""),
+_outname(""),
+_encodedName("")
+{
+    memset(&_inst, 0, sizeof(_inst));
+    unsigned int seed = static_cast<unsigned int>(0);
+    std::srand(seed);
+}
+
+void
+Test::Print()
+{
+    std::cout << _name << " completed!" << std::endl;
+    (*_log) << _name << std::endl;
+    (*_log) << _description << std::endl;
+    (*_log) << "Input file: " << _inname << std::endl;
+    (*_log) << "Output file: " << _outname << std::endl;
+    double psnr = -1.0, ssim = -1.0;
+    PSNRfromFiles(_inname.c_str(), _outname.c_str(), _inst.width, _inst.height, &psnr);
+    ssim = SSIMfromFilesMT(4 /* number of threads*/);
+
+    (*_log) << "PSNR: " << psnr << std::endl;
+    std::cout << "PSNR: " << psnr << std::endl << std::endl;
+    (*_log) << "SSIM: " << ssim << std::endl;
+    std::cout << "SSIM: " << ssim << std::endl << std::endl;
+    (*_log) << std::endl;
+}
+
+void
+Test::Setup()
+{
+    int widhei          = _inst.width*_inst.height;
+    _lengthSourceFrame  = 3*widhei/2;
+    _sourceBuffer       = new unsigned char[_lengthSourceFrame];
+}
+
+void
+Test::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
+{
+    if (bitRate > 0)
+    {
+        _bitRate = bitRate;
+    }
+    else if (_bitRate == 0)
+    {
+        _bitRate = 600;
+    }
+    _inst.maxFramerate = (unsigned char)frameRate;
+    _inst.startBitrate = (int)_bitRate;
+    _inst.maxBitrate = 8000;
+    _inst.width = width;
+    _inst.height = height;
+}
+
+void
+Test::Teardown()
+{
+    delete [] _sourceBuffer;
+}
+
+void
+Test::SetEncoder(webrtc::VideoEncoder*encoder)
+{
+    _encoder = encoder;
+}
+
+void
+Test::SetDecoder(VideoDecoder*decoder)
+{
+    _decoder = decoder;
+}
+
+void
+Test::SetLog(std::fstream* log)
+{
+    _log = log;
+}
+
+int
+Test::PSNRfromFiles(const char *refFileName, const char *testFileName, int width, int height, double *YPSNRptr)
+{
+    FILE *refFp = fopen(refFileName, "rb");
+    if( refFp == NULL ) {
+        // cannot open reference file
+        fprintf(stderr, "Cannot open file %s\n", refFileName);
+        return -1;
+    }
+
+    FILE *testFp = fopen(testFileName, "rb");
+    if( testFp == NULL ) {
+        // cannot open test file
+        fprintf(stderr, "Cannot open file %s\n", testFileName);
+        return -2;
+    }
+
+    double mse = 0.0;
+    double mseLogSum = 0.0;
+    int frames = 0;
+
+    int frameBytes = 3*width*height/2; // bytes in one frame I420
+    unsigned char *ref = new unsigned char[frameBytes]; // space for one frame I420
+    unsigned char *test = new unsigned char[frameBytes]; // space for one frame I420
+
+    int refBytes = (int) fread(ref, 1, frameBytes, refFp);
+    int testBytes = (int) fread(test, 1, frameBytes, testFp);
+
+    while( refBytes == frameBytes && testBytes == frameBytes )
+    {
+        mse = 0.0;
+
+        // calculate Y sum-square-difference
+        for( int k = 0; k < width * height; k++ )
+        {
+            mse += (test[k] - ref[k]) * (test[k] - ref[k]);
+        }
+
+        // divide by number of pixels
+        mse /= (double) (width * height);
+
+        // accumulate for total average
+        mseLogSum += std::log10( mse );
+        frames++;
+
+        refBytes = (int) fread(ref, 1, frameBytes, refFp);
+        testBytes = (int) fread(test, 1, frameBytes, testFp);
+    }
+
+    // ypsnrAvg = sum( 10 log (255^2 / MSE) ) / frames
+    //          = 20 * log(255) - 10 * mseLogSum / frames
+    *YPSNRptr = 20.0 * std::log10(255.0) - 10.0 * mseLogSum / frames;
+
+    delete [] ref;
+    delete [] test;
+
+    fclose(refFp);
+    fclose(testFp);
+
+    return 0;
+}
+int
+Test::SSIMfromFiles(const char *refFileName, const char *testFileName, int width, int height, double *SSIMptr,
+                    int startFrame /*= -1*/, int endFrame /*= -1*/)
+{
+    FILE *refFp = fopen(refFileName, "rb");
+    if( refFp == NULL ) {
+        // cannot open reference file
+        fprintf(stderr, "Cannot open file %s\n", refFileName);
+        return -1;
+    }
+
+    FILE *testFp = fopen(testFileName, "rb");
+    if( testFp == NULL ) {
+        // cannot open test file
+        fprintf(stderr, "Cannot open file %s\n", testFileName);
+        return -2;
+    }
+
+    int frames = 0;
+
+    int frameBytes = 3*width*height/2; // bytes in one frame I420
+    unsigned char *ref = new unsigned char[frameBytes]; // space for one frame I420
+    unsigned char *test = new unsigned char[frameBytes]; // space for one frame I420
+
+    if (startFrame >= 0)
+    {
+        if (fseek(refFp, frameBytes * startFrame, SEEK_SET) != 0){
+            fprintf(stderr, "Cannot go to frame %i in %s\n", startFrame, refFileName);
+            return -1;
+        }
+        if (fseek(testFp, frameBytes * startFrame, SEEK_SET) != 0){
+            fprintf(stderr, "Cannot go to frame %i in %s\n", startFrame, testFileName);
+            return -1;
+        }
+    }
+
+    int refBytes = (int) fread(ref, 1, frameBytes, refFp);
+    int testBytes = (int) fread(test, 1, frameBytes, testFp);
+
+    //
+    // SSIM: variable definition, window function, initialization
+    int window = 10;
+    int flag_window = 0;  //0 for uniform window filter, 1 for gaussian symmetric window
+    float variance_window = 2.0; //variance for window function
+    float ssimFilter[121]; //2d window filter: typically 11x11 = (window+1)*(window+1)
+    //statistics per column of window (#columns = window+1), 0 element for avg over all columns
+    float avgTest[12];
+    float avgRef[12];
+    float contrastTest[12];
+    float contrastRef[12];
+    float crossCorr[12];
+    //
+    //offsets for stability
+    float offset1 = 0.1f;
+    float offset2 = 0.1f;
+    float offset3 = offset2/2;
+    //
+    //define window for SSIM: take uniform filter for now
+    float sumfil = 0.0;
+    int nn=-1;
+    for(int j=-window/2;j<=window/2;j++)
+    for(int i=-window/2;i<=window/2;i++)
+    {
+        nn+=1;
+        if (flag_window == 0)
+            ssimFilter[nn] =  1.0;
+        else
+        {
+            float dist  = (float)(i*i) + (float)(j*j);
+            float tmp = 0.5f*dist/variance_window;
+            ssimFilter[nn] = exp(-tmp);
+        }
+        sumfil +=ssimFilter[nn];
+    }
+    //normalize window
+    nn=-1;
+    for(int j=-window/2;j<=window/2;j++)
+    for(int i=-window/2;i<=window/2;i++)
+    {
+        nn+=1;
+        ssimFilter[nn] = ssimFilter[nn]/((float)sumfil);
+    }
+    //
+    float ssimScene = 0.0; //avgerage SSIM for sequence
+    //
+    //SSIM: done with variables  and defintion
+    //
+
+    while( refBytes == frameBytes && testBytes == frameBytes &&
+        !(endFrame >= 0 && frames > endFrame - startFrame))
+    {
+        float ssimFrame = 0.0;
+        int sh = window/2+1;
+        int numPixels = 0;
+        for(int i=sh;i<height-sh;i++)
+        for(int j=sh;j<width-sh;j++)
+        {
+            avgTest[0] = 0.0;
+            avgRef[0] = 0.0;
+            contrastTest[0] = 0.0;
+            contrastRef[0] = 0.0;
+            crossCorr[0] = 0.0;
+
+            numPixels +=1;
+
+            //for uniform window, only need to loop over whole window for first column pixel in image, and then shift
+            if (j == sh || flag_window == 1)
+            {
+                //initialize statistics
+                for(int k=1;k<window+2;k++)
+                {
+                    avgTest[k] = 0.0;
+                    avgRef[k] = 0.0;
+                    contrastTest[k] = 0.0;
+                    contrastRef[k] = 0.0;
+                    crossCorr[k] = 0.0;
+                }
+                int nn=-1;
+                //compute contrast and correlation
+                for(int jj=-window/2;jj<=window/2;jj++)
+                for(int ii=-window/2;ii<=window/2;ii++)
+                {
+                    nn+=1;
+                    int i2 = i+ii;
+                    int j2 = j+jj;
+                    float tmp1 = (float)test[i2*width+j2];
+                    float tmp2 = (float)ref[i2*width+j2];
+                    //local average of each signal
+                    avgTest[jj+window/2+1] += ssimFilter[nn]*tmp1;
+                    avgRef[jj+window/2+1] += ssimFilter[nn]*tmp2;
+                    //local correlation/contrast of each signal
+                    contrastTest[jj+window/2+1] += ssimFilter[nn]*tmp1*tmp1;
+                    contrastRef[jj+window/2+1] += ssimFilter[nn]*tmp2*tmp2;
+                    //local cross correlation
+                    crossCorr[jj+window/2+1] += ssimFilter[nn]*tmp1*tmp2;
+                }
+            }
+            //for uniform window case, can shift window horiz, then compute statistics for last column in window
+            else
+            {
+                //shift statistics horiz.
+                for(int k=1;k<window+1;k++)
+                {
+                    avgTest[k]=avgTest[k+1];
+                    avgRef[k]=avgRef[k+1];
+                    contrastTest[k] = contrastTest[k+1];
+                    contrastRef[k] = contrastRef[k+1];
+                    crossCorr[k] = crossCorr[k+1];
+                }
+                //compute statistics for last column
+                avgTest[window+1] = 0.0;
+                avgRef[window+1] = 0.0;
+                contrastTest[window+1] = 0.0;
+                contrastRef[window+1] = 0.0;
+                crossCorr[window+1] = 0.0;
+                int nn = (window+1)*window - 1;
+                int jj = window/2;
+                int j2 = j + jj;
+                for(int ii=-window/2;ii<=window/2;ii++)
+                {
+                    nn+=1;
+                    int i2 = i+ii;
+                    float tmp1 = (float)test[i2*width+j2];
+                    float tmp2 = (float)ref[i2*width+j2];
+                    //local average of each signal
+                    avgTest[jj+window/2+1] += ssimFilter[nn]*tmp1;
+                    avgRef[jj+window/2+1] += ssimFilter[nn]*tmp2;
+                    //local correlation/contrast of each signal
+                    contrastTest[jj+window/2+1] += ssimFilter[nn]*tmp1*tmp1;
+                    contrastRef[jj+window/2+1] += ssimFilter[nn]*tmp2*tmp2;
+                    //local cross correlation
+                    crossCorr[jj+window/2+1] += ssimFilter[nn]*tmp1*tmp2;
+                }
+            }
+
+            //sum over all columns
+            for(int k=1;k<window+2;k++)
+            {
+                avgTest[0] += avgTest[k];
+                avgRef[0] += avgRef[k];
+                contrastTest[0] += contrastTest[k];
+                contrastRef[0] += contrastRef[k];
+                crossCorr[0] += crossCorr[k];
+            }
+
+            float tmp1 = (contrastTest[0] - avgTest[0]*avgTest[0]);
+            if (tmp1 < 0.0) tmp1 = 0.0;
+            contrastTest[0] = sqrt(tmp1);
+            float tmp2 = (contrastRef[0] - avgRef[0]*avgRef[0]);
+            if (tmp2 < 0.0) tmp2 = 0.0;
+            contrastRef[0] = sqrt(tmp2);
+            crossCorr[0] = crossCorr[0] - avgTest[0]*avgRef[0];
+
+            float ssimCorrCoeff = (crossCorr[0]+offset3)/(contrastTest[0]*contrastRef[0] + offset3);
+            float ssimLuminance = (2*avgTest[0]*avgRef[0]+offset1)/(avgTest[0]*avgTest[0] + avgRef[0]*avgRef[0] + offset1);
+            float ssimContrast   = (2*contrastTest[0]*contrastRef[0]+offset2)/(contrastTest[0]*contrastTest[0] + contrastRef[0]*contrastRef[0] + offset2);
+
+            float ssimPixel  = ssimCorrCoeff * ssimLuminance * ssimContrast;
+            ssimFrame += ssimPixel;
+        }
+        ssimFrame = ssimFrame / (numPixels);
+        //printf("***SSIM for frame ***%f \n",ssimFrame);
+        ssimScene += ssimFrame;
+        //
+        //SSIM: done with SSIM computation
+        //
+
+        frames++;
+
+        refBytes = (int) fread(ref, 1, frameBytes, refFp);
+        testBytes = (int) fread(test, 1, frameBytes, testFp);
+
+    }
+
+    //SSIM: normalize/average for sequence
+    ssimScene  = ssimScene / frames;
+    *SSIMptr = ssimScene;
+
+    delete [] ref;
+    delete [] test;
+
+    fclose(refFp);
+    fclose(testFp);
+
+    return 0;
+}
+
+bool
+Test::SSIMthread(void *vctx)
+{
+    SSIMcontext *ctx = (SSIMcontext *) vctx;
+    SSIMfromFiles(ctx->refFileName, ctx->testFileName, ctx->width, ctx->height, ctx->SSIMptr, ctx->startFrame, ctx->endFrame);
+    ctx->evnt->Set();
+    return false;
+}
+
+double Test::SSIMfromFilesMT(const int numThreads)
+{
+    int numFrames = filesize(_inname.c_str()) / _lengthSourceFrame;
+    std::vector<int> nFramesVec(numThreads);
+    std::vector<double> ssimVec(numThreads);
+    int framesPerCore = (numFrames + numThreads - 1) / numThreads; // rounding up
+    int i = 0;
+    int nFrames;
+    for (nFrames = numFrames; nFrames >= framesPerCore; nFrames -= framesPerCore)
+    {
+        nFramesVec[i++] = framesPerCore;
+    }
+    if (nFrames > 0)
+    {
+        assert(i == numThreads - 1);
+        nFramesVec[i] = nFrames; // remainder
+    }
+
+    int frameIx = 0;
+    std::vector<EventWrapper*> eventVec(numThreads);
+    std::vector<ThreadWrapper*> threadVec(numThreads);
+    std::vector<SSIMcontext> ctxVec(numThreads);
+    for (i = 0; i < numThreads; i++)
+    {
+        eventVec[i] = EventWrapper::Create();
+        ctxVec[i] = SSIMcontext(_inname.c_str(), _outname.c_str(), _inst.width, _inst.height, &ssimVec[i], frameIx, frameIx + nFramesVec[i] - 1, eventVec[i]);
+        threadVec[i] = ThreadWrapper::CreateThread(SSIMthread, &(ctxVec[i]), kLowPriority);
+        unsigned int id;
+        threadVec[i]->Start(id);
+        frameIx += nFramesVec[i];
+    }
+
+    // wait for all events
+    for (i = 0; i < numThreads; i++) {
+        eventVec[i]->Wait(100000 /* ms*/);
+        threadVec[i]->Stop();
+        delete threadVec[i];
+        delete eventVec[i];
+    }
+
+    double avgSsim = 0;
+    for (i = 0; i < numThreads; i++)
+    {
+        avgSsim += (ssimVec[i] * nFramesVec[i]);
+    }
+
+    avgSsim /= numFrames;
+    return avgSsim;
+}
+
+
+double Test::ActualBitRate(int nFrames)
+{
+    return 8.0 * _sumEncBytes / (nFrames / _inst.maxFramerate);
+}
+
+bool Test::PacketLoss(double lossRate)
+{
+    return RandUniform() < lossRate;
+}
+
+void
+Test::VideoBufferToRawImage(TestVideoBuffer& videoBuffer, RawImage &image)
+{
+    image._buffer = videoBuffer.GetBuffer();
+    image._size = videoBuffer.GetSize();
+    image._length = videoBuffer.GetLength();
+    image._width = videoBuffer.GetWidth();
+    image._height = videoBuffer.GetHeight();
+    image._timeStamp = videoBuffer.GetTimeStamp();
+}
+void
+Test::VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer, EncodedImage &image)
+{
+    image._buffer = videoBuffer.GetBuffer();
+    image._length = videoBuffer.GetLength();
+    image._size = videoBuffer.GetSize();
+    image._frameType = static_cast<VideoFrameType>(videoBuffer.GetFrameType());
+    image._timeStamp = videoBuffer.GetTimeStamp();
+    image._encodedWidth = videoBuffer.GetCaptureWidth();
+    image._encodedHeight = videoBuffer.GetCaptureHeight();
+    image._completeFrame = true;
+}
+
+long filesize(const char *filename)
+{
+FILE *f = fopen(filename,"rb");  /* open the file in read only */
+
+long size = 0;
+  if (fseek(f,0,SEEK_END)==0) /* seek was successful */
+      size = ftell(f);
+  fclose(f);
+  return size;
+}

modules/video_coding/codecs/test_framework/test.h

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
+
+#include "video_codec_interface.h"
+#include "video_buffer.h"
+#include <string>
+#include <fstream>
+#include <cstdlib>
+
+class Test
+{
+public:
+    Test(std::string name, std::string description);
+    Test(std::string name, std::string description, WebRtc_UWord32 bitRate);
+    virtual ~Test() {};
+    virtual void Perform()=0;
+    virtual void Print();
+    void SetEncoder(webrtc::VideoEncoder *encoder);
+    void SetDecoder(webrtc::VideoDecoder *decoder);
+    void SetLog(std::fstream* log);
+
+protected:
+    virtual void Setup();
+    virtual void CodecSettings(int width,
+                               int height,
+                               WebRtc_UWord32 frameRate=30,
+                               WebRtc_UWord32 bitRate=0);
+    virtual void Teardown();
+    static int PSNRfromFiles(const char *refFileName,
+                             const char *testFileName,
+                             int width,
+                             int height,
+                             double *YPSNRptr);
+    static int SSIMfromFiles(const char *refFileName,
+                             const char *testFileName,
+                             int width,
+                             int height,
+                             double *SSIMptr,
+        int startByte = -1, int endByte = -1);
+    double SSIMfromFilesMT(int numThreads);
+    static bool SSIMthread(void *ctx);
+
+    double ActualBitRate(int nFrames);
+    static bool PacketLoss(double lossRate);
+    static double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
+    static void VideoBufferToRawImage(TestVideoBuffer& videoBuffer,
+                                      webrtc::RawImage &image);
+    static void VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer,
+                                                 webrtc::EncodedImage &image);
+
+    webrtc::VideoEncoder*   _encoder;
+    webrtc::VideoDecoder*   _decoder;
+    WebRtc_UWord32          _bitRate;
+    unsigned int            _lengthSourceFrame;
+    unsigned char*          _sourceBuffer;
+    TestVideoBuffer         _inputVideoBuffer;
+    TestVideoEncodedBuffer  _encodedVideoBuffer;
+    TestVideoBuffer         _decodedVideoBuffer;
+    webrtc::VideoCodec      _inst;
+    std::fstream*           _log;
+    std::string             _inname;
+    std::string             _outname;
+    std::string             _encodedName;
+    int                     _sumEncBytes;
+
+private:
+    std::string             _name;
+    std::string             _description;
+
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_

modules/video_coding/codecs/test_framework/test_framework.gyp

@@ -0,0 +1,61 @@
+# Copyright (c) 2009 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+  'includes': [
+    '../../../../common_settings.gypi', # Common settings
+  ],
+  'targets': [
+    {
+      'target_name': 'test_framework',
+      'type': '<(library)',
+
+      'dependencies': [
+        '../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '../../../../common_video/vplib/main/source/vplib.gyp:webrtc_vplib',
+      ],
+
+      'include_dirs': [
+        '../interface',
+      ],
+
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '../interface',
+        ],
+      },
+
+      'sources': [
+        # header files
+        'benchmark.h',
+        'normal_async_test.h',
+        'normal_test.h',
+        'packet_loss_test.h',
+        'performance_test.h',
+        'test.h',
+        'unit_test.h',
+        'video_buffer.h',
+        'video_source.h',
+
+        # source files
+        'benchmark.cc',
+        'normal_async_test.cc',
+        'normal_test.cc',
+        'packet_loss_test.cc',
+        'performance_test.cc',
+        'test.cc',
+        'unit_test.cc',
+        'video_buffer.cc',
+        'video_source.cc',
+
+      ],
+    },
+  ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:

modules/video_coding/codecs/test_framework/unit_test.cc

@@ -0,0 +1,815 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "unit_test.h"
+#include "video_source.h"
+#include "tick_util.h"
+#include <cassert>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+using namespace webrtc;
+
+UnitTest::UnitTest()
+:
+Test("UnitTest", "Unit test"),
+_tests(0),
+_errors(0),
+_source(NULL),
+_refFrame(NULL),
+_refEncFrame(NULL),
+_refDecFrame(NULL),
+_refEncFrameLength(0),
+_sourceFile(NULL),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL)
+{
+}
+
+UnitTest::UnitTest(std::string name, std::string description)
+:
+_tests(0),
+_errors(0),
+_source(NULL),
+_refFrame(NULL),
+_refEncFrame(NULL),
+_refDecFrame(NULL),
+_refEncFrameLength(0),
+_sourceFile(NULL),
+_encodeCompleteCallback(NULL),
+_decodeCompleteCallback(NULL),
+Test(name, description)
+{
+}
+
+UnitTest::~UnitTest()
+{
+    if (_encodeCompleteCallback) {
+        delete _encodeCompleteCallback;
+    }
+
+    if (_decodeCompleteCallback) {
+        delete _decodeCompleteCallback;
+    }
+
+    if (_source) {
+        delete _source;
+    }
+
+    if (_refFrame) {
+        delete [] _refFrame;
+    }
+
+    if (_refDecFrame) {
+        delete [] _refDecFrame;
+    }
+
+    if (_sourceBuffer) {
+        delete [] _sourceBuffer;
+    }
+
+    if (_sourceFile) {
+        fclose(_sourceFile);
+    }
+
+    if (_refEncFrame) {
+        delete [] _refEncFrame;
+    }
+}
+
+WebRtc_Word32
+UnitTestEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
+                                        const void* codecSpecificInfo,
+                                        const webrtc::RTPFragmentationHeader*
+                                        fragmentation)
+{
+    _encodedVideoBuffer->VerifyAndAllocate(encodedImage._size);
+    _encodedVideoBuffer->CopyBuffer(encodedImage._size, encodedImage._buffer);
+    _encodedVideoBuffer->UpdateLength(encodedImage._length);
+    _encodedVideoBuffer->SetFrameType(encodedImage._frameType);
+    _encodedVideoBuffer->SetCaptureWidth(
+        (WebRtc_UWord16)encodedImage._encodedWidth);
+    _encodedVideoBuffer->SetCaptureHeight(
+        (WebRtc_UWord16)encodedImage._encodedHeight);
+    _encodedVideoBuffer->SetTimeStamp(encodedImage._timeStamp);
+    _encodeComplete = true;
+    _encodedFrameType = encodedImage._frameType;
+    return 0;
+}
+
+WebRtc_Word32 UnitTestDecodeCompleteCallback::Decoded(RawImage& image)
+{
+    _decodedVideoBuffer->VerifyAndAllocate(image._length);
+    _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
+    _decodedVideoBuffer->SetWidth(image._width);
+    _decodedVideoBuffer->SetHeight(image._height);
+    _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
+    _decodeComplete = true;
+    return 0;
+}
+
+bool
+UnitTestEncodeCompleteCallback::EncodeComplete()
+{
+    if (_encodeComplete)
+    {
+        _encodeComplete = false;
+        return true;
+    }
+    return false;
+}
+
+VideoFrameType
+UnitTestEncodeCompleteCallback::EncodedFrameType() const
+{
+    return _encodedFrameType;
+}
+
+bool
+UnitTestDecodeCompleteCallback::DecodeComplete()
+{
+    if (_decodeComplete)
+    {
+        _decodeComplete = false;
+        return true;
+    }
+    return false;
+}
+
+WebRtc_UWord32
+UnitTest::WaitForEncodedFrame() const
+{
+    WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+    while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs)
+    {
+        if (_encodeCompleteCallback->EncodeComplete())
+        {
+            return _encodedVideoBuffer.GetLength();
+        }
+    }
+    return 0;
+}
+
+WebRtc_UWord32
+UnitTest::WaitForDecodedFrame() const
+{
+    WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
+    while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitDecTimeMs)
+    {
+        if (_decodeCompleteCallback->DecodeComplete())
+        {
+            return _decodedVideoBuffer.GetLength();
+        }
+    }
+    return 0;
+}
+
+WebRtc_UWord32
+UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
+                                   WebRtc_UWord32 /* frameRate */)
+{
+    return _encoder->SetRates(bitRate, _inst.maxFramerate);
+}
+
+void
+UnitTest::Setup()
+{
+    // Use _sourceFile as a check to prevent multiple Setup() calls.
+    if (_sourceFile != NULL)
+    {
+        return;
+    }
+
+    if (_encodeCompleteCallback == NULL)
+    {
+        _encodeCompleteCallback =
+            new UnitTestEncodeCompleteCallback(&_encodedVideoBuffer);
+    }
+    if (_decodeCompleteCallback == NULL)
+    {
+        _decodeCompleteCallback =
+            new UnitTestDecodeCompleteCallback(&_decodedVideoBuffer);
+    }
+
+    _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
+    _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
+
+    _source = new VideoSource("test/testFiles/foreman_cif.yuv", kCIF);
+
+    _lengthSourceFrame = _source->GetFrameLength();
+    _refFrame = new unsigned char[_lengthSourceFrame];
+    _refDecFrame = new unsigned char[_lengthSourceFrame];
+    _sourceBuffer = new unsigned char [_lengthSourceFrame];
+    _sourceFile = fopen(_source->GetFileName().c_str(), "rb");
+    VIDEO_TEST_EXIT_ON_ERR(_sourceFile != NULL);
+
+    _inst.maxFramerate = _source->GetFrameRate();
+    _bitRate = 300;
+    _inst.startBitrate = 300;
+    _inst.maxBitrate = 4000;
+    _inst.width = _source->GetWidth();
+    _inst.height = _source->GetHeight();
+
+    // Get input frame.
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    VIDEO_TEST_EXIT_ON_ERR(fread(_refFrame, 1, _lengthSourceFrame, _sourceFile)
+                           == _lengthSourceFrame);
+    _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+    rewind(_sourceFile);
+
+    // Get a reference encoded frame.
+    _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+
+    RawImage image;
+    VideoBufferToRawImage(_inputVideoBuffer, image);
+
+    // Ensures our initial parameters are valid.
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    _encoder->Encode(image, NULL);
+    _refEncFrameLength = WaitForEncodedFrame();
+    VIDEO_TEST_EXIT_ON_ERR(_refEncFrameLength > 0);
+    _refEncFrame = new unsigned char[_refEncFrameLength];
+    memcpy(_refEncFrame, _encodedVideoBuffer.GetBuffer(), _refEncFrameLength);
+
+    // Get a reference decoded frame.
+    _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+
+    int frameLength = 0;
+    int i=0;
+    while (frameLength == 0)
+    {
+        if (i > 0)
+        {
+            // Insert yet another frame
+            _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+            VIDEO_TEST_EXIT_ON_ERR(fread(_refFrame, 1, _lengthSourceFrame,
+                _sourceFile) == _lengthSourceFrame);
+            _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+            _inputVideoBuffer.SetWidth(_source->GetWidth());
+            _inputVideoBuffer.SetHeight(_source->GetHeight());
+            VideoBufferToRawImage(_inputVideoBuffer, image);
+            _encoder->Encode(image, NULL);
+            VIDEO_TEST_EXIT_ON_ERR(WaitForEncodedFrame() > 0);
+        }
+        EncodedImage encodedImage;
+        VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+        VIDEO_TEST_EXIT_ON_ERR(_decoder->Decode(encodedImage, 0, NULL)
+                               == WEBRTC_VIDEO_CODEC_OK);
+        frameLength = WaitForDecodedFrame();
+        _encodedVideoBuffer.Reset();
+        _encodedVideoBuffer.UpdateLength(0);
+        i++;
+    }
+    rewind(_sourceFile);
+    VIDEO_TEST(frameLength == _lengthSourceFrame);
+    memcpy(_refDecFrame, _decodedVideoBuffer.GetBuffer(), _lengthSourceFrame);
+}
+
+void
+UnitTest::Teardown()
+{
+    // Use _sourceFile as a check to prevent multiple Teardown() calls.
+    if (_sourceFile == NULL)
+    {
+        return;
+    }
+
+    _encoder->Release();
+    _decoder->Release();
+
+    fclose(_sourceFile);
+    _sourceFile = NULL;
+    delete [] _refFrame;
+    _refFrame = NULL;
+    delete [] _refEncFrame;
+    _refEncFrame = NULL;
+    delete [] _refDecFrame;
+    _refDecFrame = NULL;
+    delete [] _sourceBuffer;
+    _sourceBuffer = NULL;
+}
+
+void
+UnitTest::Print()
+{
+    printf("Unit Test\n\n%i tests completed\n", _tests);
+    if (_errors > 0)
+    {
+        printf("%i FAILED\n\n", _errors);
+    }
+    else
+    {
+        printf("ALL PASSED\n\n");
+    }
+}
+
+int
+UnitTest::DecodeWithoutAssert()
+{
+    EncodedImage encodedImage;
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    int ret = _decoder->Decode(encodedImage, 0, NULL);
+    int frameLength = WaitForDecodedFrame();
+    _encodedVideoBuffer.Reset();
+    _encodedVideoBuffer.UpdateLength(0);
+    return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
+}
+
+int
+UnitTest::Decode()
+{
+    EncodedImage encodedImage;
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    if (encodedImage._length == 0)
+    {
+        return WEBRTC_VIDEO_CODEC_OK;
+    }
+    int ret = _decoder->Decode(encodedImage, 0, NULL);
+    int frameLength = WaitForDecodedFrame();
+    assert(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
+        == _lengthSourceFrame));
+    VIDEO_TEST(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
+        == _lengthSourceFrame));
+    _encodedVideoBuffer.Reset();
+    _encodedVideoBuffer.UpdateLength(0);
+    return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
+}
+
+// Test pure virtual VideoEncoder and VideoDecoder APIs.
+void
+UnitTest::Perform()
+{
+    UnitTest::Setup();
+    int frameLength;
+    RawImage inputImage;
+    EncodedImage encodedImage;
+    EventWrapper& sleepEvent = *EventWrapper::Create();
+
+    //----- Encoder parameter tests -----
+
+    //-- Calls before InitEncode() --
+    // We want to revert the initialization done in Setup().
+    VIDEO_TEST(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+    VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+    VIDEO_TEST(_encoder->Encode(inputImage, NULL)
+               == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+    VIDEO_TEST(_encoder->Reset() == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+    //-- InitEncode() errors --
+    // Null pointer.
+    VIDEO_TEST(_encoder->InitEncode(NULL, 1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    // bit rate exceeds max bit rate
+    WebRtc_Word32 tmpBitRate = _inst.startBitrate;
+    WebRtc_Word32 tmpMaxBitRate = _inst.maxBitrate;
+    _inst.startBitrate = 4000;
+    _inst.maxBitrate = 3000;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440)  ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _inst.startBitrate = tmpBitRate;
+    _inst.maxBitrate = tmpMaxBitRate; //unspecified value
+
+    // Bad framerate.
+    _inst.maxFramerate = 0;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    // Seems like we should allow any framerate in range [0, 255].
+    //_inst.frameRate = 100;
+    //VIDEO_TEST(_encoder->InitEncode(&_inst, 1) == -1); // FAILS
+    _inst.maxFramerate = 30;
+
+    // Bad bitrate.
+    _inst.startBitrate = -1;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _inst.maxBitrate = _inst.startBitrate - 1;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _inst.maxBitrate = 0;
+    _inst.startBitrate = 300;
+
+    // Bad maxBitRate.
+    _inst.maxBitrate = 200;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _inst.maxBitrate = 4000;
+
+    // Bad width.
+    _inst.width = 0;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) < 0);
+    // Should there be a width and height cap?
+    //_inst.width = 10000;
+    //VIDEO_TEST(_encoder->InitEncode(&_inst, 1) == -1);
+    _inst.width = _source->GetWidth();
+
+    // Bad height.
+    _inst.height = 0;
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) < 0);
+    _inst.height = _source->GetHeight();
+
+    // Bad number of cores.
+    VIDEO_TEST(_encoder->InitEncode(&_inst, -1, 1440) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+
+    //-- Encode() errors --
+
+    // inputVideoBuffer unallocated.
+    _inputVideoBuffer.Free();
+    VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+    VIDEO_TEST(_encoder->Encode(inputImage, NULL) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
+
+    //----- Encoder stress tests -----
+
+    // Vary frame rate and I-frame request.
+    VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+    for (int i = 1; i <= 60; i++)
+    {
+        VideoFrameType frameType = !(i % 2) ? kKeyFrame : kDeltaFrame;
+        VIDEO_TEST(_encoder->Encode(inputImage, NULL, frameType) ==
+            WEBRTC_VIDEO_CODEC_OK);
+        VIDEO_TEST(WaitForEncodedFrame() > 0);
+        sleepEvent.Wait(10); // Allow the encoder's queue to realize it's empty.
+    }
+
+    // Init then encode.
+    _encodedVideoBuffer.UpdateLength(0);
+    _encodedVideoBuffer.Reset();
+    VIDEO_TEST(_encoder->Encode(inputImage, NULL) == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(WaitForEncodedFrame() > 0);
+
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    _encoder->Encode(inputImage, NULL);
+    frameLength = WaitForEncodedFrame();
+    VIDEO_TEST(frameLength > 0);
+    VIDEO_TEST(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+            _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+    // Reset then encode.
+    _encodedVideoBuffer.UpdateLength(0);
+    _encodedVideoBuffer.Reset();
+    VIDEO_TEST(_encoder->Encode(inputImage, NULL) == WEBRTC_VIDEO_CODEC_OK);
+    WaitForEncodedFrame();
+    VIDEO_TEST(_encoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    _encoder->Encode(inputImage, NULL);
+    frameLength = WaitForEncodedFrame();
+    VIDEO_TEST(frameLength > 0);
+    VIDEO_TEST(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+        _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+    // Release then encode.
+    _encodedVideoBuffer.UpdateLength(0);
+    _encodedVideoBuffer.Reset();
+    VIDEO_TEST(_encoder->Encode(inputImage, NULL) == WEBRTC_VIDEO_CODEC_OK);
+    WaitForEncodedFrame();
+    VIDEO_TEST(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    _encoder->Encode(inputImage, NULL);
+    frameLength = WaitForEncodedFrame();
+    VIDEO_TEST(frameLength > 0);
+    VIDEO_TEST(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
+        _encodedVideoBuffer.GetBuffer(), frameLength) == true);
+
+    //----- Decoder parameter tests -----
+
+    //-- Calls before InitDecode() --
+    // We want to revert the initialization done in Setup().
+    VIDEO_TEST(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    VIDEO_TEST(_decoder->Decode(encodedImage, false, NULL) ==
+        WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+    WaitForDecodedFrame();
+    VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+
+    //-- Decode() errors --
+    // Unallocated encodedVideoBuffer.
+    _encodedVideoBuffer.Free();
+    //_encodedVideoBuffer.UpdateLength(10); // Buffer NULL but length > 0
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    VIDEO_TEST(_decoder->Decode(encodedImage, false, NULL) ==
+        WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
+    _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+
+    //----- Decoder stress tests -----
+    unsigned char* tmpBuf = new unsigned char[_lengthSourceFrame];
+
+    // "Random" and zero data.
+    // We either expect an error, or at the least, no output.
+    // This relies on the codec's ability to detect an erroneous bitstream.
+    /*
+    VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+    for (int i = 0; i < 100; i++)
+    {
+        VIDEO_TEST_EXIT_ON_ERR(fread(tmpBuf, 1, _refEncFrameLength, _sourceFile)
+            == _refEncFrameLength);
+        _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
+        VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+        FillDecoderSpecificInfo(encodedImage);
+        int ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
+        VIDEO_TEST(ret <= 0);
+        if (ret == 0)
+        {
+            VIDEO_TEST(WaitForDecodedFrame() == 0);
+        }
+
+        memset(tmpBuf, 0, _refEncFrameLength);
+        _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
+        VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+        FillDecoderSpecificInfo(encodedImage);
+        ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
+        VIDEO_TEST(ret <= 0);
+        if (ret == 0)
+        {
+            VIDEO_TEST(WaitForDecodedFrame() == 0);
+        }
+    }
+    */
+    rewind(_sourceFile);
+
+    _encodedVideoBuffer.UpdateLength(_refEncFrameLength);
+    _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, _refEncFrame);
+
+    // Init then decode.
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+    frameLength = 0;
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    while (frameLength == 0)
+    {
+        _decoder->Decode(encodedImage, false, NULL);
+        frameLength = WaitForDecodedFrame();
+    }
+    VIDEO_TEST(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+        _refDecFrame, _lengthSourceFrame) == true);
+
+    // Reset then decode.
+    VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    frameLength = 0;
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    while (frameLength == 0)
+    {
+        _decoder->Decode(encodedImage, false, NULL);
+        frameLength = WaitForDecodedFrame();
+    }
+    VIDEO_TEST(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+        _refDecFrame, _lengthSourceFrame) == true);
+
+    // Decode with other size, reset, then decode with original size again
+    // to verify that decoder is reset to a "fresh" state upon Reset().
+    {
+        // assert that input frame size is a factor of two, so that we can use
+        // quarter size below
+        VIDEO_TEST((_inst.width % 2 == 0) && (_inst.height % 2 == 0));
+
+        VideoCodec tempInst;
+        memcpy(&tempInst, &_inst, sizeof(VideoCodec));
+        tempInst.width /= 2;
+        tempInst.height /= 2;
+
+        // Encode reduced (quarter) frame size
+        VIDEO_TEST(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+        VIDEO_TEST(_encoder->InitEncode(&tempInst, 1, 1440) ==
+            WEBRTC_VIDEO_CODEC_OK);
+        RawImage tempInput(inputImage._buffer, inputImage._length/4,
+                           inputImage._size/4);
+        _encoder->Encode(tempInput, NULL);
+        frameLength = WaitForEncodedFrame();
+        VIDEO_TEST(frameLength > 0);
+
+        // Reset then decode.
+        VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+        frameLength = 0;
+        VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+        while (frameLength == 0)
+        {
+            _decoder->Decode(encodedImage, false, NULL);
+            frameLength = WaitForDecodedFrame();
+        }
+
+        // Encode original frame again
+        VIDEO_TEST(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+        VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) ==
+            WEBRTC_VIDEO_CODEC_OK);
+        _encoder->Encode(inputImage, NULL);
+        frameLength = WaitForEncodedFrame();
+        VIDEO_TEST(frameLength > 0);
+
+        // Reset then decode original frame again.
+        VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+        frameLength = 0;
+        VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+        while (frameLength == 0)
+        {
+            _decoder->Decode(encodedImage, false, NULL);
+            frameLength = WaitForDecodedFrame();
+        }
+
+        // check that decoded frame matches with reference
+        VIDEO_TEST(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+            _refDecFrame, _lengthSourceFrame) == true);
+
+    }
+
+    // Release then decode.
+    VIDEO_TEST(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+    frameLength = 0;
+    VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
+    while (frameLength == 0)
+    {
+        _decoder->Decode(encodedImage, false, NULL);
+        frameLength = WaitForDecodedFrame();
+    }
+    VIDEO_TEST(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
+        _refDecFrame, _lengthSourceFrame) == true);
+    _encodedVideoBuffer.UpdateLength(0);
+    _encodedVideoBuffer.Reset();
+
+    delete [] tmpBuf;
+
+    //----- Function tests -----
+    int frames = 0;
+    // Do not specify maxBitRate (as in ViE).
+    _inst.maxBitrate = 0;
+
+    //-- Timestamp propagation --
+    VIDEO_TEST(_encoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+
+    printf("\nTimestamp propagation test...\n");
+    frames = 0;
+    int frameDelay = 0;
+    int encTimeStamp;
+    _decodedVideoBuffer.SetTimeStamp(0);
+    while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+        _lengthSourceFrame)
+    {
+        _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+        _inputVideoBuffer.SetTimeStamp(frames);
+        VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+        VIDEO_TEST_EXIT_ON_ERR(_encoder->Encode(inputImage, NULL) ==
+            WEBRTC_VIDEO_CODEC_OK);
+        frameLength = WaitForEncodedFrame();
+        //VIDEO_TEST_EXIT_ON_ERR(frameLength);
+        VIDEO_TEST(frameLength > 0);
+        encTimeStamp = _encodedVideoBuffer.GetTimeStamp();
+        VIDEO_TEST(_inputVideoBuffer.GetTimeStamp() == encTimeStamp);
+
+        frameLength = Decode();
+        if (frameLength == 0)
+        {
+            frameDelay++;
+        }
+
+        encTimeStamp -= frameDelay;
+        if (encTimeStamp < 0)
+        {
+            encTimeStamp = 0;
+        }
+        VIDEO_TEST(_decodedVideoBuffer.GetTimeStamp() == encTimeStamp);
+        frames++;
+        sleepEvent.Wait(33);
+    }
+    delete &sleepEvent;
+    VIDEO_TEST_EXIT_ON_ERR(feof(_sourceFile) != 0);
+    rewind(_sourceFile);
+
+    RateControlTests();
+
+    Teardown();
+}
+
+void
+UnitTest::RateControlTests()
+{
+    FILE *outFile = NULL;
+    std::string outFileName;
+    int frames = 0;
+    RawImage inputImage;
+    WebRtc_UWord32 frameLength;
+    EventWrapper& sleepEvent = *EventWrapper::Create();
+
+    // Do not specify maxBitRate (as in ViE).
+    _inst.maxBitrate = 0;
+    //-- Verify rate control --
+    VIDEO_TEST(_encoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+    // add: should also be 0, and 1
+    const int bitRate[] =
+    {100, 200, 300, 400, 500, 600, 800, 1000, 2000, 3000, 4000, 10000};
+    const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
+
+    printf("\nRate control test\n");
+    for (int i = 0; i < nBitrates; i++)
+    {
+        _bitRate = bitRate[i];
+        int totalBytes = 0;
+        _encoder->Reset();
+        _inst.startBitrate = _bitRate;
+        _encoder->InitEncode(&_inst, 4, 1440);
+        _decoder->Reset();
+        _decoder->InitDecode(&_inst, 1);
+        frames = 0;
+
+        if (_bitRate > _inst.maxBitrate)
+        {
+            CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
+        }
+        else
+        {
+            CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
+        }
+
+        while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
+            _lengthSourceFrame)
+        {
+            _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
+            _inputVideoBuffer.SetTimeStamp(_inputVideoBuffer.GetTimeStamp() +
+                static_cast<WebRtc_UWord32>(9e4 /
+                    static_cast<float>(_inst.maxFramerate)));
+            VideoBufferToRawImage(_inputVideoBuffer, inputImage);
+            VIDEO_TEST_EXIT_ON_ERR(_encoder->Encode(inputImage, NULL) ==
+                WEBRTC_VIDEO_CODEC_OK);
+            frameLength = WaitForEncodedFrame();
+            VIDEO_TEST_EXIT_ON_ERR(frameLength > 0);
+            //VIDEO_TEST(frameLength > 0);
+            totalBytes += frameLength;
+            frames++;
+
+            _encodedVideoBuffer.UpdateLength(0);
+            _encodedVideoBuffer.Reset();
+
+            sleepEvent.Wait(10);
+        }
+        WebRtc_UWord32 actualBitrate =
+            (totalBytes  / frames * _inst.maxFramerate * 8)/1000;
+        printf("Target bitrate: %d kbps, actual bitrate: %d kbps\n", _bitRate,
+            actualBitrate);
+        // Test for close match over reasonable range.
+        if (_bitRate >= 100 && _bitRate <= 4000)
+        {
+            //VIDEO_TEST(fabs(actualBitrate - _bitRate) < 0.05 * _bitRate);
+            VIDEO_TEST(abs(WebRtc_Word32(actualBitrate - _bitRate)) <
+                0.1 * _bitRate); // for VP8
+        }
+        VIDEO_TEST_EXIT_ON_ERR(feof(_sourceFile) != 0);
+        rewind(_sourceFile);
+    }
+}
+
+bool
+UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+                          const void* ptrB, unsigned int bLengthBytes)
+{
+    if (aLengthBytes != bLengthBytes)
+    {
+        return false;
+    }
+
+    return memcmp(ptrA, ptrB, aLengthBytes) == 0;
+}

modules/video_coding/codecs/test_framework/unit_test.h

@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+
+#include "test.h"
+#include "event_wrapper.h"
+
+// Disable "conditional expression is constant" warnings on the perfectly
+// acceptable
+// do { ... } while (0) constructions below.
+// Refer to http://stackoverflow.com/questions/1946445/
+//   is-there-better-way-to-write-do-while0-construct-to-avoid-compiler-warnings
+// for some discussion of the issue.
+#pragma warning(disable : 4127)
+
+#define VIDEO_TEST(expr)                                                       \
+    do                                                                         \
+    {                                                                          \
+        _tests++;                                                              \
+        if (!(expr))                                                           \
+        {                                                                      \
+            fprintf(stderr, "Error at line %i of %s\nAssertion failed: %s\n\n",\
+                __LINE__, __FILE__, #expr);                                    \
+            _errors++;                                                         \
+        }                                                                      \
+    } while (0)
+
+#define VIDEO_TEST_EXIT_ON_ERR(expr)                                           \
+    do                                                                         \
+    {                                                                          \
+        if (!(expr))                                                           \
+        {                                                                      \
+            fprintf(stderr, "Error at line %i of %s\nAssertion failed: %s\n",  \
+                __LINE__, __FILE__, #expr);                                    \
+            fprintf(stderr, "Exiting...\n\n");                                 \
+            exit(EXIT_FAILURE);                                                \
+        }                                                                      \
+    } while (0)
+
+class VideoSource;
+class UnitTestEncodeCompleteCallback;
+class UnitTestDecodeCompleteCallback;
+
+class UnitTest : public Test
+{
+public:
+    UnitTest();
+    virtual ~UnitTest();
+    virtual void Perform();
+    virtual void Print();
+
+protected:
+    UnitTest(std::string name, std::string description);
+    virtual WebRtc_UWord32 CodecSpecific_SetBitrate(
+        WebRtc_UWord32 bitRate,
+        WebRtc_UWord32 /* frameRate */);
+    virtual void Setup();
+    virtual void Teardown();
+    virtual void RateControlTests();
+    virtual int Decode();
+    virtual int DecodeWithoutAssert();
+    virtual int SetCodecSpecificParameters() {return 0;};
+
+    virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+                                 const void *ptrB, unsigned int bLengthBytes);
+
+    WebRtc_UWord32 WaitForEncodedFrame() const;
+    WebRtc_UWord32 WaitForDecodedFrame() const;
+
+    int _tests;
+    int _errors;
+
+    VideoSource* _source;
+    unsigned char* _refFrame;
+    unsigned char* _refEncFrame;
+    unsigned char* _refDecFrame;
+    int _refEncFrameLength;
+    FILE* _sourceFile;
+
+    UnitTestEncodeCompleteCallback* _encodeCompleteCallback;
+    UnitTestDecodeCompleteCallback* _decodeCompleteCallback;
+    enum { kMaxWaitEncTimeMs = 100 };
+    enum { kMaxWaitDecTimeMs = 25 };
+};
+
+class UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback
+{
+public:
+    UnitTestEncodeCompleteCallback(TestVideoEncodedBuffer* buffer,
+                                   WebRtc_UWord32 decoderSpecificSize = 0,
+                                   void* decoderSpecificInfo = NULL) :
+      _encodedVideoBuffer(buffer),
+      _decoderSpecificSize(decoderSpecificSize),
+      _decoderSpecificInfo(decoderSpecificInfo),
+      _encodeComplete(false) {}
+    WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage,
+                          const void* codecSpecificInfo,
+                          const webrtc::RTPFragmentationHeader*
+                          fragmentation = NULL);
+    bool EncodeComplete();
+    // Note that this only makes sense if an encode has been completed
+    webrtc::VideoFrameType EncodedFrameType() const;
+private:
+    TestVideoEncodedBuffer* _encodedVideoBuffer;
+    void* _decoderSpecificInfo;
+    WebRtc_UWord32 _decoderSpecificSize;
+    bool _encodeComplete;
+    webrtc::VideoFrameType _encodedFrameType;
+};
+
+class UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+    UnitTestDecodeCompleteCallback(TestVideoBuffer* buffer) :
+        _decodedVideoBuffer(buffer), _decodeComplete(false) {}
+    WebRtc_Word32 Decoded(webrtc::RawImage& image);
+    bool DecodeComplete();
+private:
+    TestVideoBuffer* _decodedVideoBuffer;
+    bool _decodeComplete;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
+

modules/video_coding/codecs/test_framework/video_buffer.cc

@@ -0,0 +1,319 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <string.h>
+#include "video_buffer.h"
+
+using namespace webrtc;
+
+TestVideoBuffer::TestVideoBuffer():
+_buffer(0),
+_bufferSize(0),
+_bufferLength(0),
+_startOffset(0),
+_timeStamp(0),
+_width(0),
+_height(0)
+{
+   //
+}
+
+
+TestVideoBuffer::~TestVideoBuffer()
+{
+    _timeStamp = 0;
+    _startOffset = 0;
+    _bufferLength = 0;
+    _bufferSize = 0;
+
+    if(_buffer)
+    {
+        delete [] _buffer;
+        _buffer = 0;
+    }
+}
+
+TestVideoBuffer::TestVideoBuffer(const TestVideoBuffer& rhs)
+:
+_buffer(0),
+_bufferSize(rhs._bufferSize),
+_bufferLength(rhs._bufferLength),
+_startOffset(rhs._startOffset),
+_timeStamp(rhs._timeStamp),
+_width(rhs._width),
+_height(rhs._height)
+{
+    // make sure that our buffer is big enough
+    _buffer = new unsigned char[_bufferSize];
+
+    // only copy required length
+    memcpy(_buffer + _startOffset, rhs._buffer, _bufferLength);  // GetBuffer() includes _startOffset
+}
+
+void TestVideoBuffer::SetTimeStamp(unsigned int timeStamp)
+{
+    _timeStamp = timeStamp;
+}
+
+unsigned int
+TestVideoBuffer::GetWidth() const
+{
+    return _width;
+}
+
+unsigned int
+TestVideoBuffer::GetHeight() const
+{
+    return _height;
+}
+
+void
+TestVideoBuffer::SetWidth(unsigned int width)
+{
+    _width = width;
+}
+
+void
+TestVideoBuffer::SetHeight(unsigned int height)
+{
+    _height = height;
+}
+
+
+void TestVideoBuffer::Free()
+{
+    _timeStamp = 0;
+    _startOffset = 0;
+    _bufferLength = 0;
+    _bufferSize = 0;
+    _height = 0;
+    _width = 0;
+
+    if(_buffer)
+    {
+        delete [] _buffer;
+        _buffer = 0;
+    }
+}
+
+void TestVideoBuffer::VerifyAndAllocate(unsigned int minimumSize)
+{
+    if(minimumSize > _bufferSize)
+    {
+        // make sure that our buffer is big enough
+        unsigned char * newBufferBuffer = new unsigned char[minimumSize];
+        if(_buffer)
+        {
+            // copy the old data
+            memcpy(newBufferBuffer, _buffer, _bufferSize);
+            delete [] _buffer;
+        }
+        _buffer = newBufferBuffer;
+        _bufferSize = minimumSize;
+    }
+}
+
+int TestVideoBuffer::SetOffset(unsigned int length)
+{
+    if (length > _bufferSize ||
+        length > _bufferLength)
+    {
+        return -1;
+    }
+
+    unsigned int oldOffset = _startOffset;
+
+    if(oldOffset > length)
+    {
+        unsigned int newLength = _bufferLength + (oldOffset-length);// increase by the diff
+        assert(newLength <= _bufferSize);
+        _bufferLength = newLength;
+    }
+    if(oldOffset < length)
+    {
+        if(_bufferLength > (length-oldOffset))
+        {
+            _bufferLength -= (length-oldOffset); // decrease by the diff
+        }
+    }
+    _startOffset = length; // update
+
+    return 0;
+}
+
+void TestVideoBuffer::UpdateLength(unsigned int newLength)
+{
+    assert(newLength +_startOffset <= _bufferSize);
+    _bufferLength = newLength;
+}
+
+void TestVideoBuffer::CopyBuffer(unsigned int length, const unsigned char* buffer)
+{
+    assert(length+_startOffset <= _bufferSize);
+    memcpy(_buffer+_startOffset, buffer, length);
+    _bufferLength = length;
+}
+
+void TestVideoBuffer::CopyBuffer(TestVideoBuffer& fromVideoBuffer)
+{
+    assert(fromVideoBuffer.GetLength() + fromVideoBuffer.GetStartOffset() <= _bufferSize);
+    assert(fromVideoBuffer.GetSize() <= _bufferSize);
+
+    _bufferLength = fromVideoBuffer.GetLength();
+    _startOffset = fromVideoBuffer.GetStartOffset();
+    _timeStamp = fromVideoBuffer.GetTimeStamp();
+    _height = fromVideoBuffer.GetHeight();
+    _width = fromVideoBuffer.GetWidth();
+
+    // only copy required length
+    memcpy(_buffer+_startOffset, fromVideoBuffer.GetBuffer(), fromVideoBuffer.GetLength());  // GetBuffer() includes _startOffset
+
+}
+
+void TestVideoBuffer::CopyPointer(const TestVideoBuffer& fromVideoBuffer)
+{
+    _bufferSize = fromVideoBuffer.GetSize();
+    _bufferLength = fromVideoBuffer.GetLength();
+    _startOffset = fromVideoBuffer.GetStartOffset();
+    _timeStamp = fromVideoBuffer.GetTimeStamp();
+    _height = fromVideoBuffer.GetHeight();
+    _width = fromVideoBuffer.GetWidth();
+
+    _buffer = fromVideoBuffer.GetBuffer();
+}
+
+void TestVideoBuffer::ClearPointer()
+{
+    _buffer = NULL;
+}
+
+void TestVideoBuffer::SwapBuffers(TestVideoBuffer& videoBuffer)
+{
+    unsigned char*  tempBuffer = _buffer;
+    unsigned int    tempSize = _bufferSize;
+    unsigned int    tempLength =_bufferLength;
+    unsigned int    tempOffset = _startOffset;
+    unsigned int    tempTime = _timeStamp;
+    unsigned int    tempHeight = _height;
+    unsigned int    tempWidth = _width;
+
+    _buffer = videoBuffer.GetBuffer();
+    _bufferSize = videoBuffer.GetSize();
+    _bufferLength = videoBuffer.GetLength();
+    _startOffset = videoBuffer.GetStartOffset();
+    _timeStamp =  videoBuffer.GetTimeStamp();
+    _height = videoBuffer.GetHeight();
+    _width = videoBuffer.GetWidth();
+
+
+    videoBuffer.Set(tempBuffer, tempSize, tempLength, tempOffset, tempTime);
+    videoBuffer.SetHeight(tempHeight);
+    videoBuffer.SetWidth(tempWidth);
+}
+
+void TestVideoBuffer::Set(unsigned char* tempBuffer,unsigned int tempSize,unsigned int tempLength, unsigned int tempOffset,unsigned int timeStamp)
+{
+    _buffer = tempBuffer;
+    _bufferSize = tempSize;
+    _bufferLength = tempLength;
+    _startOffset = tempOffset;
+    _timeStamp = timeStamp;
+}
+
+unsigned char* TestVideoBuffer::GetBuffer() const
+{
+    return _buffer+_startOffset;
+}
+
+unsigned int TestVideoBuffer::GetStartOffset() const
+{
+    return _startOffset;
+}
+
+unsigned int TestVideoBuffer::GetSize() const
+{
+    return _bufferSize;
+}
+
+unsigned int TestVideoBuffer::GetLength() const
+{
+    return _bufferLength;
+}
+
+unsigned int TestVideoBuffer::GetTimeStamp() const
+{
+    return _timeStamp;
+}
+
+/**
+*   TestVideoEncodedBuffer
+*
+*/
+
+TestVideoEncodedBuffer::TestVideoEncodedBuffer() :
+    _captureWidth(0),
+    _captureHeight(0),
+    _frameRate(-1)
+{
+    _frameType = kDeltaFrame;
+}
+
+TestVideoEncodedBuffer::~TestVideoEncodedBuffer()
+{
+}
+
+void TestVideoEncodedBuffer::SetCaptureWidth(unsigned short width)
+{
+    _captureWidth = width;
+}
+
+void TestVideoEncodedBuffer::SetCaptureHeight(unsigned short height)
+{
+    _captureHeight = height;
+}
+
+unsigned short TestVideoEncodedBuffer::GetCaptureWidth()
+{
+    return _captureWidth;
+}
+
+unsigned short TestVideoEncodedBuffer::GetCaptureHeight()
+{
+    return _captureHeight;
+}
+
+VideoFrameType TestVideoEncodedBuffer::GetFrameType()
+{
+    return _frameType;
+}
+
+void TestVideoEncodedBuffer::SetFrameType(VideoFrameType frametype)
+{
+    _frameType = frametype;
+}
+
+void TestVideoEncodedBuffer::Reset()
+{
+    _captureWidth = 0;
+    _captureHeight = 0;
+    _frameRate = -1;
+    _frameType = kDeltaFrame;
+}
+
+void  TestVideoEncodedBuffer::SetFrameRate(float frameRate)
+{
+    _frameRate = frameRate;
+}
+
+float  TestVideoEncodedBuffer::GetFrameRate()
+{
+    return _frameRate;
+}

modules/video_coding/codecs/test_framework/video_buffer.h

@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
+
+#include "typedefs.h"
+#include "video_image.h"
+
+class TestVideoBuffer
+{
+public:
+    TestVideoBuffer();
+
+    virtual ~TestVideoBuffer();
+
+    TestVideoBuffer(const TestVideoBuffer& rhs);
+
+    /**
+    * Verifies that current allocated buffer size is larger than or equal to the input size.
+    * If the current buffer size is smaller, a new allocation is made and the old buffer data is copied to the new buffer.
+    */
+    void VerifyAndAllocate(unsigned int minimumSize);
+
+    void UpdateLength(unsigned int newLength);
+
+    void SwapBuffers(TestVideoBuffer& videoBuffer);
+
+    void CopyBuffer(unsigned int length, const unsigned char* fromBuffer);
+
+    void CopyBuffer(TestVideoBuffer& fromVideoBuffer);
+
+    // Use with care, and remember to call ClearPointer() when done.
+    void CopyPointer(const TestVideoBuffer& fromVideoBuffer);
+
+    void ClearPointer();
+
+    int  SetOffset(unsigned int length);            // Sets offset to beginning of frame in buffer
+
+    void Free();                                    // Deletes frame buffer and resets members to zero
+
+    void SetTimeStamp(unsigned int timeStamp);      // Sets timestamp of frame (90kHz)
+
+    /**
+    *   Gets pointer to frame buffer
+    */
+    unsigned char* GetBuffer() const;
+
+    /**
+    *   Gets allocated buffer size
+    */
+    unsigned int	GetSize() const;
+
+    /**
+    *   Gets length of frame
+    */
+    unsigned int	GetLength() const;
+
+    /**
+    *   Gets timestamp of frame (90kHz)
+    */
+    unsigned int	GetTimeStamp() const;
+
+    unsigned int	GetWidth() const;
+    unsigned int	GetHeight() const;
+
+    void            SetWidth(unsigned int width);
+    void            SetHeight(unsigned int height);
+
+private:
+    TestVideoBuffer& operator=(const TestVideoBuffer& inBuffer);
+
+private:
+    void Set(unsigned char* buffer,unsigned int size,unsigned int length,unsigned int offset, unsigned int timeStamp);
+    unsigned int GetStartOffset() const;
+
+    unsigned char*		  _buffer;          // Pointer to frame buffer
+    unsigned int		  _bufferSize;      // Allocated buffer size
+    unsigned int		  _bufferLength;    // Length (in bytes) of frame
+    unsigned int		  _startOffset;     // Offset (in bytes) to beginning of frame in buffer
+    unsigned int		  _timeStamp;       // Timestamp of frame (90kHz)
+    unsigned int          _width;
+    unsigned int          _height;
+};
+
+class TestVideoEncodedBuffer: public TestVideoBuffer
+{
+public:
+    TestVideoEncodedBuffer();
+    ~TestVideoEncodedBuffer();
+
+    void SetCaptureWidth(unsigned short width);
+    void SetCaptureHeight(unsigned short height);
+    unsigned short GetCaptureWidth();
+    unsigned short GetCaptureHeight();
+
+    webrtc::VideoFrameType GetFrameType();
+    void SetFrameType(webrtc::VideoFrameType frametype);
+
+    void Reset();
+
+    void SetFrameRate(float frameRate);
+    float GetFrameRate();
+
+private:
+    TestVideoEncodedBuffer& operator=(const TestVideoEncodedBuffer& inBuffer);
+
+private:
+    unsigned short			   _captureWidth;
+    unsigned short			   _captureHeight;
+    webrtc::VideoFrameType     _frameType;
+    float                      _frameRate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_

modules/video_coding/codecs/test_framework/video_source.cc

@@ -0,0 +1,417 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_source.h"
+#include "vplib.h"
+#include <cassert>
+#include <stdio.h>
+
+VideoSource::VideoSource()
+:
+_fileName("../../../../testFiles/foreman.yuv"),
+_width(352),
+_height(288),
+_type(webrtc::kI420),
+_frameRate(30)
+{
+}
+
+VideoSource::VideoSource(std::string fileName, VideoSize size,
+    int frameRate /*= 30*/, webrtc::VideoType type /*=  webrtc::kI420*/)
+:
+_fileName(fileName),
+_type(type),
+_frameRate(frameRate)
+{
+    assert(size != kUndefined && size != kNumberOfVideoSizes);
+    assert(type != webrtc::kUnknown);
+    assert(frameRate > 0);
+    assert(GetWidthHeight(size, _width, _height) == 0);
+}
+
+VideoSource::VideoSource(std::string fileName, int width, int height,
+    int frameRate /*= 30*/,  webrtc::VideoType type /*=  webrtc::kI420*/)
+:
+_fileName(fileName),
+_width(width),
+_height(height),
+_type(type),
+_frameRate(frameRate)
+{
+    assert(width > 0);
+    assert(height > 0);
+    assert(type != webrtc::kUnknown);
+    assert(frameRate > 0);
+}
+
+VideoSize
+VideoSource::GetSize() const
+{
+    return GetSize(_width, _height);
+}
+
+VideoSize
+VideoSource::GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height)
+{
+    if(width == 128 && height == 96)
+    {
+        return kSQCIF;
+    }else if(width == 160 && height == 120)
+    {
+        return kQQVGA;
+    }else if(width == 176 && height == 144)
+    {
+        return kQCIF;
+    }else if(width == 320 && height == 240)
+    {
+        return kQVGA;
+    }else if(width == 352 && height == 288)
+    {
+        return kCIF;
+    }else if(width == 640 && height == 480)
+    {
+        return kVGA;
+    }else if(width == 720 && height == 480)
+    {
+        return kNTSC;
+    }else if(width == 704 && height == 576)
+    {
+        return k4CIF;
+    }else if(width == 800 && height == 600)
+    {
+        return kSVGA;
+    }else if(width == 960 && height == 720)
+    {
+        return kHD;
+    }else if(width == 1024 && height == 768)
+    {
+        return kXGA;
+    }else if(width == 1440 && height == 1080)
+    {
+        return kFullHD;
+    }else if(width == 400 && height == 240)
+    {
+        return kWQVGA;
+    }else if(width == 800 && height == 480)
+    {
+        return kWVGA;
+    }else if(width == 1280 && height == 720)
+    {
+        return kWHD;
+    }else if(width == 1920 && height == 1080)
+    {
+        return kWFullHD;
+    }
+    return kUndefined;
+}
+
+unsigned int
+VideoSource::GetFrameLength() const
+{
+    return webrtc::CalcBufferSize(_type, _width, _height);
+}
+
+const char*
+VideoSource::GetMySizeString() const
+{
+    return VideoSource::GetSizeString(GetSize());
+}
+
+const char*
+VideoSource::GetSizeString(VideoSize size)
+{
+    switch (size)
+    {
+        case kSQCIF:
+            return "SQCIF";
+        case kQQVGA:
+            return "QQVGA";
+        case kQCIF:
+            return "QCIF";
+        case kQVGA:
+            return "QVGA";
+        case kCIF:
+            return "CIF";
+        case kVGA:
+            return "VGA";
+        case kNTSC:
+            return "NTSC";
+        case k4CIF:
+            return "4CIF";
+        case kSVGA:
+            return "SVGA";
+        case kHD:
+            return "HD";
+        case kXGA:
+            return "XGA";
+        case kFullHD:
+            return "Full_HD";
+        case kWQVGA:
+            return "WQVGA";
+        case kWHD:
+            return "WHD";
+        case kWFullHD:
+            return "WFull_HD";
+        default:
+            return "Undefined";
+    }
+}
+
+std::string
+VideoSource::GetFilePath() const
+{
+    size_t slashPos = _fileName.find_last_of("/\\");
+    if (slashPos == std::string::npos)
+    {
+        return ".";
+    }
+
+    return _fileName.substr(0, slashPos);
+}
+
+std::string
+VideoSource::GetName() const
+{
+    // Remove path.
+    size_t slashPos = _fileName.find_last_of("/\\");
+    if (slashPos == std::string::npos)
+    {
+        slashPos = 0;
+    }
+    else
+    {
+        slashPos++;
+    }
+
+    // Remove extension and underscored suffix if it exists.
+    return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
+        _fileName.find_last_of(".")) - slashPos);
+}
+
+void
+VideoSource::Convert(const VideoSource &target, bool force /* = false */) const
+{
+    // Ensure target rate is less than or equal to source
+    // (i.e. we are only temporally downsampling).
+    assert(target.GetFrameRate() <= _frameRate);
+    // Only supports YUV420 currently.
+    assert(_type == webrtc::kI420 && target.GetType() == webrtc::kI420);
+    if (!force && (FileExists(target.GetFileName().c_str()) ||
+        (target.GetWidth() == _width && target.GetHeight() == _height && target.GetFrameRate() == _frameRate)))
+    {
+        // Assume that the filename uniquely defines the content.
+        // If the file already exists, it is the correct file.
+        return;
+    }
+    FILE *inFile = NULL;
+    FILE *outFile = NULL;
+
+    inFile = fopen(_fileName.c_str(), "rb");
+    assert(inFile != NULL);
+
+    outFile = fopen(target.GetFileName().c_str(), "wb");
+    assert(outFile != NULL);
+
+    FrameDropper fd;
+    fd.SetFrameRate(target.GetFrameRate(), _frameRate);
+
+    const size_t lengthOutFrame = webrtc::CalcBufferSize(target.GetType(),
+        target.GetWidth(), target.GetHeight());
+    assert(lengthOutFrame > 0);
+    unsigned char *outFrame = new unsigned char[lengthOutFrame];
+
+    const size_t lengthInFrame = webrtc::CalcBufferSize(_type, _width, _height);
+    assert(lengthInFrame > 0);
+    unsigned char *inFrame = new unsigned char[lengthInFrame];
+
+    while (fread(inFrame, 1, lengthInFrame, inFile) == lengthInFrame)
+    {
+        if (!fd.DropFrame())
+        {
+            assert(target.GetWidth() == _width &&
+                   target.GetHeight() == _height); // Add video interpolator here!
+            fwrite(outFrame, 1, lengthOutFrame, outFile);
+        }
+    }
+
+    delete inFrame;
+    delete outFrame;
+    fclose(inFile);
+    fclose(outFile);
+}
+
+bool VideoSource::FileExists(const char* fileName)
+{
+    FILE* fp = NULL;
+    fp = fopen(fileName, "rb");
+    if(fp != NULL)
+    {
+        fclose(fp);
+        return true;
+    }
+    return false;
+}
+
+
+int
+VideoSource::GetWidthHeight( VideoSize size, int & width, int& height)
+{
+    switch(size)
+    {
+    case kSQCIF:
+        width = 128;
+        height = 96;
+        return 0;
+    case kQQVGA:
+        width = 160;
+        height = 120;
+        return 0;
+    case kQCIF:
+        width = 176;
+        height = 144;
+        return 0;
+    case kCGA:
+        width = 320;
+        height = 200;
+        return 0;
+    case kQVGA:
+        width = 320;
+        height = 240;
+        return 0;
+    case kSIF:
+        width = 352;
+        height = 240;
+        return 0;
+    case kWQVGA:
+        width = 400;
+        height = 240;
+        return 0;
+    case kCIF:
+        width = 352;
+        height = 288;
+        return 0;
+    case kW288p:
+        width = 512;
+        height = 288;
+        return 0;
+    case k448p:
+        width = 576;
+        height = 448;
+        return 0;
+    case kVGA:
+        width = 640;
+        height = 480;
+        return 0;
+    case k432p:
+        width = 720;
+        height = 432;
+        return 0;
+    case kW432p:
+        width = 768;
+        height = 432;
+        return 0;
+    case k4SIF:
+        width = 704;
+        height = 480;
+        return 0;
+    case kW448p:
+        width = 768;
+        height = 448;
+        return 0;
+    case kNTSC:
+        width = 720;
+        height = 480;
+        return 0;
+    case kFW448p:
+        width = 800;
+        height = 448;
+        return 0;
+    case kWVGA:
+        width = 800;
+        height = 480;
+        return 0;
+    case k4CIF:
+        width = 704;
+        height = 576;
+        return 0;
+    case kSVGA:
+        width = 800;
+        height = 600;
+        return 0;
+    case kW544p:
+        width = 960;
+        height = 544;
+        return 0;
+    case kW576p:
+        width = 1024;
+        height = 576;
+        return 0;
+    case kHD:
+        width = 960;
+        height = 720;
+        return 0;
+    case kXGA:
+        width = 1024;
+        height = 768;
+        return 0;
+    case kFullHD:
+        width = 1440;
+        height = 1080;
+        return 0;
+    case kWHD:
+        width = 1280;
+        height = 720;
+        return 0;
+    case kWFullHD:
+        width = 1920;
+        height = 1080;
+        return 0;
+    default:
+        return -1;
+    }
+}
+
+FrameDropper::FrameDropper()
+:
+_dropsBetweenRenders(0),
+_frameCounter(0)
+{
+}
+
+bool
+FrameDropper::DropFrame()
+{
+    _frameCounter++;
+    if (_frameCounter > _dropsBetweenRenders)
+    {
+        _frameCounter = 0;
+        return false;
+    }
+    return true;
+}
+
+unsigned int
+FrameDropper::DropsBetweenRenders()
+{
+    return _dropsBetweenRenders;
+}
+
+void
+FrameDropper::SetFrameRate(double frameRate, double maxFrameRate)
+{
+    if (frameRate >= 1.0)
+    {
+        _dropsBetweenRenders = static_cast<unsigned int>(maxFrameRate / frameRate + 0.5) - 1;
+    }
+    else
+    {
+        _dropsBetweenRenders = 0;
+    }
+}

modules/video_coding/codecs/test_framework/video_source.h

@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+
+#include <string>
+
+#include "vplib.h"
+
+enum VideoSize
+    {
+        kUndefined,
+        kSQCIF,     // 128*96       = 12 288
+        kQQVGA,     // 160*120      = 19 200
+        kQCIF,      // 176*144      = 25 344
+        kCGA,       // 320*200      = 64 000
+        kQVGA,      // 320*240      = 76 800
+        kSIF,       // 352*240      = 84 480
+        kWQVGA,     // 400*240      = 96 000
+        kCIF,       // 352*288      = 101 376
+        kW288p,     // 512*288      = 147 456 (WCIF)
+        k448p,      // 576*448      = 281 088
+        kVGA,       // 640*480      = 307 200
+        k432p,      // 720*432      = 311 040
+        kW432p,     // 768*432      = 331 776
+        k4SIF,      // 704*480      = 337 920
+        kW448p,     // 768*448      = 344 064
+        kNTSC,		// 720*480      = 345 600
+        kFW448p,    // 800*448      = 358 400
+        kWVGA,      // 800*480      = 384 000
+        k4CIF,      // 704�576      = 405 504
+        kSVGA,      // 800*600      = 480 000
+        kW544p,     // 960*544      = 522 240
+        kW576p,     // 1024*576     = 589 824 (W4CIF)
+        kHD,        // 960*720      = 691 200
+        kXGA,       // 1024*768     = 786 432
+        kWHD,       // 1280*720     = 921 600
+        kFullHD,    // 1440*1080    = 1 555 200
+        kWFullHD,   // 1920*1080    = 2 073 600
+
+        kNumberOfVideoSizes
+    };
+
+class VideoSource
+{
+public:
+    VideoSource();
+    VideoSource(std::string fileName, VideoSize size, int frameRate = 30,
+        webrtc::VideoType type = webrtc::kI420);
+    VideoSource(std::string fileName, int width, int height, int frameRate = 30,
+                webrtc::VideoType type = webrtc::kI420);
+
+    std::string GetFileName() const { return _fileName; }
+    int GetWidth() const { return _width; }
+    int GetHeight() const { return _height; }
+    webrtc::VideoType GetType() const { return _type; }
+    int GetFrameRate() const { return _frameRate; }
+
+    // Returns the file path without a trailing slash.
+    std::string GetFilePath() const;
+
+    // Returns the filename with the path (including the leading slash) removed.
+    std::string GetName() const;
+
+    VideoSize GetSize() const;
+    static VideoSize GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
+    unsigned int GetFrameLength() const;
+
+    // Returns a human-readable size string.
+    static const char* GetSizeString(VideoSize size);
+    const char* GetMySizeString() const;
+
+    // Opens the video source, converting and writing to the specified target.
+    // If force is true, the conversion will be done even if the target file
+    // already exists.
+    void Convert(const VideoSource& target, bool force = false) const;
+    static bool FileExists(const char* fileName);
+private:
+    static int GetWidthHeight( VideoSize size, int& width, int& height);
+    std::string _fileName;
+    int _width;
+    int _height;
+    webrtc::VideoType _type;
+    int _frameRate;
+};
+
+class FrameDropper
+{
+public:
+    FrameDropper();
+    bool DropFrame();
+    unsigned int DropsBetweenRenders();
+    void SetFrameRate(double frameRate, double maxFrameRate);
+
+private:
+    unsigned int _dropsBetweenRenders;
+    unsigned int _frameCounter;
+};
+
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
+

modules/video_coding/codecs/vp8/main/interface/vp8.h

@@ -0,0 +1,223 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * vp8.h
+ * WEBRTC VP8 wrapper interface
+ */
+
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
+
+#include "video_codec_interface.h"
+
+// VPX forward declaration
+typedef struct vpx_codec_ctx vpx_codec_ctx_t;
+typedef struct vpx_codec_ctx vpx_dec_ctx_t;
+typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
+typedef struct vpx_image vpx_image_t;
+
+namespace webrtc
+{
+
+/******************************/
+/* VP8Encoder class           */
+/******************************/
+class VP8Encoder : public VideoEncoder
+{
+public:
+    VP8Encoder();
+    virtual ~VP8Encoder();
+
+// Free encoder memory.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 Release();
+
+// Reset encoder state and prepare for a new call.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+//                               <0 - Errors:
+//                                 WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+//                                 WEBRTC_VIDEO_CODEC_ERROR
+    virtual WebRtc_Word32 Reset();
+
+// Initialize the encoder with the information from the codecSettings
+//
+// Input:
+//          - codecSettings     : Codec settings
+//          - numberOfCores     : Number of cores available for the encoder
+//          - maxPayloadSize    : The maximum size each payload is allowed
+//                                to have. Usually MTU - overhead.
+//
+// Return value                 : Set bit rate if OK
+//                                <0 - Errors:
+//                                  WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+//                                  WEBRTC_VIDEO_CODEC_ERR_SIZE
+//                                  WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED
+//                                  WEBRTC_VIDEO_CODEC_MEMORY
+//                                  WEBRTC_VIDEO_CODEC_ERROR
+    virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings,
+                                     WebRtc_Word32 numberOfCores,
+                                     WebRtc_UWord32 maxPayloadSize);
+
+// Encode an I420 image (as a part of a video stream). The encoded image
+// will be returned to the user through the encode complete callback.
+//
+// Input:
+//          - inputImage        : Image to be encoded
+//          - frameTypes        : Frame type to be generated by the encoder.
+//
+// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+//                                <0 - Errors:
+//                                  WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+//                                  WEBRTC_VIDEO_CODEC_MEMORY
+//                                  WEBRTC_VIDEO_CODEC_ERROR
+//                                  WEBRTC_VIDEO_CODEC_TIMEOUT
+
+    virtual WebRtc_Word32 Encode(const RawImage& inputImage,
+                                 const void* codecSpecificInfo,
+                                 VideoFrameType frameType);
+
+// Register an encode complete callback object.
+//
+// Input:
+//          - callback         : Callback object which handles encoded images.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
+
+// Inform the encoder of the new packet loss rate in the network
+//
+//          - packetLoss       : Fraction lost
+//                               (loss rate in percent = 100 * packetLoss / 255)
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK
+//                               <0 - Errors:
+//                                  WEBRTC_VIDEO_CODEC_ERROR
+//
+    virtual WebRtc_Word32 SetPacketLoss(WebRtc_UWord32 packetLoss);
+
+// Inform the encoder about the new target bit rate.
+//
+//          - newBitRate       : New target bit rate
+//          - frameRate        : The target frame rate
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRateKbit, WebRtc_UWord32 frameRate);
+
+// Get version number for the codec.
+//
+// Input:
+//      - version       : Pointer to allocated char buffer.
+//      - buflen        : Length of provided char buffer.
+//
+// Output:
+//      - version       : Version number string written to char buffer.
+//
+// Return value         : >0 - Length of written string.
+//                        <0 - WEBRTC_VIDEO_CODEC_ERR_SIZE
+    virtual WebRtc_Word32 Version(WebRtc_Word8 *version, WebRtc_Word32 length) const;
+    static WebRtc_Word32  VersionStatic(WebRtc_Word8 *version, WebRtc_Word32 length);
+
+private:
+    EncodedImage              _encodedImage;
+    EncodedImageCallback*     _encodedCompleteCallback;
+    WebRtc_Word32             _width;
+    WebRtc_Word32             _height;
+    WebRtc_Word32             _maxBitRateKbit;
+    int                       _maxFrameRate;
+    bool                      _inited;
+    WebRtc_UWord16            _pictureID;
+    bool                      _pictureLossIndicationOn;
+    bool                      _feedbackModeOn;
+    bool                      _nextRefIsGolden;
+    bool                      _lastAcknowledgedIsGolden;
+    bool                      _haveReceivedAcknowledgement;
+    WebRtc_UWord16            _pictureIDLastSentRef;
+    WebRtc_UWord16            _pictureIDLastAcknowledgedRef;
+
+    vpx_codec_ctx_t*          _encoder;
+    vpx_codec_enc_cfg_t*      _cfg;
+    vpx_image_t*              _raw;
+};// end of VP8Encoder class
+
+/******************************/
+/* VP8Decoder class           */
+/******************************/
+class VP8Decoder : public VideoDecoder
+{
+public:
+    VP8Decoder();
+    virtual ~VP8Decoder();
+
+// Initialize the decoder.
+//
+// Return value         :  WEBRTC_VIDEO_CODEC_OK.
+//                        <0 - Errors:
+//                                  WEBRTC_VIDEO_CODEC_ERROR
+    virtual WebRtc_Word32 InitDecode(const VideoCodec* inst,
+                                     WebRtc_Word32 numberOfCores);
+
+// Decode encoded image (as a part of a video stream). The decoded image
+// will be returned to the user through the decode complete callback.
+//
+// Input:
+//          - inputImage        : Encoded image to be decoded
+//          - missingFrames     : True if one or more frames have been lost
+//                                since the previous decode call.
+//          - codecSpecificInfo : pointer to specific codec data
+//          - renderTimeMs      : Render time in Ms
+//
+// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+//                                <0 - Errors:
+//                                      WEBRTC_VIDEO_CODEC_ERROR
+//                                      WEBRTC_VIDEO_CODEC_ERR_PARAMETER
+    virtual WebRtc_Word32 Decode(const EncodedImage& inputImage,
+                                 bool missingFrames,
+                                 const void* /*codecSpecificInfo*/,
+                                 WebRtc_Word64 /*renderTimeMs*/);
+
+// Register a decode complete callback object.
+//
+// Input:
+//          - callback         : Callback object which handles decoded images.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+    virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
+
+// Free decoder memory.
+//
+// Return value                : WEBRTC_VIDEO_CODEC_OK if OK
+//                               <0 - Errors:
+//                                      WEBRTC_VIDEO_CODEC_ERROR
+    virtual WebRtc_Word32 Release();
+
+// Reset decoder state and prepare for a new call.
+//
+// Return value         : WEBRTC_VIDEO_CODEC_OK.
+//                        <0 - Errors:
+//                                  WEBRTC_VIDEO_CODEC_UNINITIALIZED
+//                                  WEBRTC_VIDEO_CODEC_ERROR
+    virtual WebRtc_Word32 Reset();
+    virtual WebRtc_Word32 SetCodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return -1; }
+
+private:
+    RawImage                   _decodedImage;
+    DecodedImageCallback*      _decodeCompleteCallback;
+    bool                       _inited;
+    bool                       _feedbackModeOn;
+    vpx_dec_ctx_t*             _decoder;
+
+};// end of VP8Decoder class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_

modules/video_coding/codecs/vp8/main/source/vp8.cc

@@ -0,0 +1,806 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * vp8.cc
+ *
+ * This file contains the WEBRTC VP8 wrapper implementation
+ *
+ */
+#include "vp8.h"
+#include "tick_util.h"
+
+#include "vpx/vpx_encoder.h"
+#include "vpx/vpx_decoder.h"
+#include "vpx/vp8cx.h"
+#include "vpx/vp8dx.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+
+#include "module_common_types.h"
+
+#define VP8_FREQ_HZ 90000
+//#define DEV_PIC_LOSS
+
+namespace webrtc
+{
+
+VP8Encoder::VP8Encoder():
+    _encodedImage(),
+    _encodedCompleteCallback(NULL),
+    _width(0),
+    _height(0),
+    _maxBitRateKbit(0),
+    _inited(false),
+    _pictureID(0),
+    _pictureLossIndicationOn(false),
+    _feedbackModeOn(false),
+    _nextRefIsGolden(true),
+    _lastAcknowledgedIsGolden(true),
+    _haveReceivedAcknowledgement(false),
+    _pictureIDLastSentRef(0),
+    _pictureIDLastAcknowledgedRef(0),
+    _encoder(NULL),
+    _cfg(NULL),
+    _raw(NULL)
+{
+    srand((WebRtc_UWord32)TickTime::MillisecondTimestamp());
+}
+
+VP8Encoder::~VP8Encoder()
+{
+    Release();
+}
+
+WebRtc_Word32
+VP8Encoder::VersionStatic(WebRtc_Word8* version, WebRtc_Word32 length)
+{
+    const WebRtc_Word8* str = "WebM/VP8 version 1.0.0\n"; // Bali
+    WebRtc_Word32 verLen = (WebRtc_Word32)strlen(str);
+    if (verLen > length)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    strncpy(version, str, length);
+    return verLen;
+}
+
+WebRtc_Word32
+VP8Encoder::Version(WebRtc_Word8 *version, WebRtc_Word32 length) const
+{
+    return VersionStatic(version, length);
+}
+
+WebRtc_Word32
+VP8Encoder::Release()
+{
+    if (_encodedImage._buffer != NULL)
+    {
+        delete [] _encodedImage._buffer;
+        _encodedImage._buffer = NULL;
+    }
+    if (_encoder != NULL)
+    {
+        if (vpx_codec_destroy(_encoder))
+        {
+            return WEBRTC_VIDEO_CODEC_MEMORY;
+        }
+        delete _encoder;
+        _encoder = NULL;
+    }
+    if (_cfg != NULL)
+    {
+        delete _cfg;
+        _cfg = NULL;
+    }
+    if (_raw != NULL)
+    {
+        vpx_img_free(_raw);
+        delete _raw;
+        _raw = NULL;
+    }
+    _inited = false;
+
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Encoder::Reset()
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    // reinitialize encoder to initial state
+    if (vpx_codec_enc_init(_encoder, vpx_codec_vp8_cx(), _cfg, 0))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Encoder::SetRates(WebRtc_UWord32 newBitRateKbit, WebRtc_UWord32 newFrameRate)
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (_encoder->err)
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    if (newFrameRate < 1)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+
+    // update bit rate
+    if (_maxBitRateKbit > 0 && newBitRateKbit > static_cast<WebRtc_UWord32>(_maxBitRateKbit))
+    {
+        newBitRateKbit = _maxBitRateKbit;
+    }
+    _cfg->rc_target_bitrate = newBitRateKbit; // in kbit/s
+
+    // update frame rate
+    if (newFrameRate != _maxFrameRate)
+    {
+        _maxFrameRate = static_cast<int>(newFrameRate);
+        _cfg->g_timebase.num = 1;
+        _cfg->g_timebase.den = _maxFrameRate;//VP8_FREQ_HZ;
+    }
+
+    // update encoder context
+    if (vpx_codec_enc_config_set(_encoder, _cfg))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Encoder::InitEncode(const VideoCodec* inst,
+                       WebRtc_Word32 numberOfCores,
+                       WebRtc_UWord32 /*maxPayloadSize */)
+{
+    if (inst == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (inst->maxFramerate < 1)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (inst->startBitrate < 0 || inst->maxBitrate < 0)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    // allow zero to represent an unspecified maxBitRate
+    if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (inst->width < 1 || inst->height < 1)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (numberOfCores < 1)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+#ifdef DEV_PIC_LOSS
+    // we need to know if we use feedback
+    _feedbackModeOn = inst->codecSpecific.VP8.feedbackModeOn;
+    _pictureLossIndicationOn = inst->codecSpecific.VP8.pictureLossIndicationOn;
+#endif
+
+    WebRtc_Word32 retVal = Release();
+    if (retVal < 0)
+    {
+        return retVal;
+    }
+    if (_encoder == NULL)
+    {
+        _encoder = new vpx_codec_ctx_t;
+    }
+    if (_cfg == NULL)
+    {
+        _cfg = new vpx_codec_enc_cfg_t;
+    }
+    if (_raw == NULL)
+    {
+        _raw = new vpx_image_t;
+    }
+
+    _maxBitRateKbit = inst->maxBitrate;
+    _maxFrameRate = inst->maxFramerate;
+    _width = inst->width;
+    _height = inst->height;
+
+    // random start 16 bits is enough
+    _pictureID = (WebRtc_UWord16)rand();
+
+    // allocate memory for encoded image
+    if (_encodedImage._buffer != NULL)
+    {
+        delete [] _encodedImage._buffer;
+    }
+    _encodedImage._size = (3 * inst->width * inst->height) >> 1;
+    _encodedImage._buffer = new WebRtc_UWord8[_encodedImage._size];
+    if (_encodedImage._buffer == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_MEMORY;
+    }
+
+    vpx_img_alloc(_raw, IMG_FMT_I420, inst->width, inst->height, 1);
+    // populate encoder configuration with default values
+    if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), _cfg, 0))
+    {
+         return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    _cfg->g_w = inst->width;
+    _cfg->g_h = inst->height;
+    if (_maxBitRateKbit > 0 && inst->startBitrate > static_cast<unsigned int>(_maxBitRateKbit))
+    {
+        _cfg->rc_target_bitrate = _maxBitRateKbit;
+    }
+    else
+    {
+      _cfg->rc_target_bitrate = inst->startBitrate;  // in kbit/s
+    }
+
+    // setting the time base of the codec
+    _cfg->g_timebase.num = 1;
+    _cfg->g_timebase.den = _maxFrameRate;
+
+    _cfg->g_error_resilient = 1;  //enabled
+    _cfg->g_lag_in_frames = 0; // 0- no frame lagging
+
+    _cfg->g_threads = numberOfCores;
+
+    // rate control settings
+    _cfg->rc_dropframe_thresh = 0;
+    _cfg->rc_end_usage = VPX_CBR;
+    _cfg->g_pass = VPX_RC_ONE_PASS;
+    _cfg->rc_resize_allowed = 0;
+    _cfg->rc_min_quantizer = 4;
+    _cfg->rc_max_quantizer = 56;
+    _cfg->rc_undershoot_pct = 98;
+    _cfg->rc_buf_initial_sz = 500;
+    _cfg->rc_buf_optimal_sz = 600;
+    _cfg->rc_buf_sz = 1000;
+
+
+#ifdef DEV_PIC_LOSS
+    // this can only be off if we know we use feedback
+    if (_pictureLossIndicationOn)
+    {
+        _cfg->kf_mode = VPX_KF_DISABLED; // don't generate key frame unless we tell you
+    }
+    else
+#endif
+    {
+        _cfg->kf_mode = VPX_KF_AUTO;
+        _cfg->kf_max_dist = 300;
+    }
+
+    // construct encoder context
+    if (vpx_codec_enc_init(_encoder, vpx_codec_vp8_cx(), _cfg, 0))
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    switch (inst->codecSpecific.VP8.complexity)
+    {
+        case kComplexityHigh:
+        {
+            vpx_codec_control(_encoder, VP8E_SET_CPUUSED, -5);
+            break;
+        }
+        case kComplexityHigher:
+        {
+            vpx_codec_control(_encoder, VP8E_SET_CPUUSED, -4);
+            break;
+        }
+        case kComplexityMax:
+        {
+            vpx_codec_control(_encoder, VP8E_SET_CPUUSED, -3);
+           break;
+        }
+        default:
+        {
+            vpx_codec_control(_encoder, VP8E_SET_CPUUSED, -6);
+            break;
+        }
+    }
+
+    _inited = true;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Encoder::Encode(const RawImage& inputImage,
+                             const void* codecSpecificInfo,
+                             VideoFrameType frameTypes)
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (inputImage._buffer == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (_encodedCompleteCallback == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+
+    vpx_codec_iter_t iter = NULL;
+
+    // image in vpx_image_t format
+    _raw->planes[PLANE_Y] =  inputImage._buffer;
+    _raw->planes[PLANE_U] =  &inputImage._buffer[_height * _width];
+    _raw->planes[PLANE_V] =  &inputImage._buffer[_height * _width * 5 >> 2];
+
+    int flags = 0;
+    if (frameTypes == kKeyFrame)
+    {
+        flags |= VPX_EFLAG_FORCE_KF; // will update both golden and altref
+        _encodedImage._frameType = kKeyFrame;
+        _pictureIDLastSentRef = _pictureID;
+    }
+    else
+    {
+#ifdef DEV_PIC_LOSS
+        if (_feedbackModeOn && codecSpecificInfo)
+        {
+            const CodecSpecificInfo* info = static_cast<const CodecSpecificInfo*>(codecSpecificInfo);
+            if (info->codecType == kVideoCodecVP8)
+            {
+                // codecSpecificInfo will contain received RPSI and SLI picture IDs
+                // this will help us decide on when to switch type of reference frame
+
+                // if we receive SLI
+                // force using an old golden or altref as a reference
+
+                if (info->codecSpecific.VP8.hasReceivedSLI)
+                {
+                    // if this is older than my last acked ref we can ignore it
+                    // info->codecSpecific.VP8.pictureIdSLI valid 6 bits => 64 frames
+
+                    // since picture id can wrap check if in between our last sent and last acked
+
+                    bool sendRefresh = false;
+                    // check for a wrap in picture ID
+                    if ((_pictureIDLastAcknowledgedRef & 0x3f) > (_pictureID & 0x3f))
+                    {
+                        // we have a wrap
+                        if ( info->codecSpecific.VP8.pictureIdSLI > (_pictureIDLastAcknowledgedRef&0x3f)||
+                            info->codecSpecific.VP8.pictureIdSLI < (_pictureID & 0x3f))
+                        {
+                            sendRefresh = true;
+                        }
+                    }
+                    else if (info->codecSpecific.VP8.pictureIdSLI > (_pictureIDLastAcknowledgedRef&0x3f)&&
+                             info->codecSpecific.VP8.pictureIdSLI < (_pictureID & 0x3f))
+                    {
+                        sendRefresh = true;
+                    }
+
+                    // right now we could also ignore it if it's older than our last sent ref since
+                    // last sent ref only refers back to last acked
+                    // _pictureIDLastSentRef;
+                    if (sendRefresh)
+                    {
+                        flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame
+
+                        if (_haveReceivedAcknowledgement)
+                        {
+                            // we cant set this if we refer to a key frame
+                            if (_lastAcknowledgedIsGolden)
+                            {
+                                flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alternate reference frame
+                            }
+                            else
+                            {
+                                flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame
+                            }
+                        }
+                    }
+                }
+                if (info->codecSpecific.VP8.hasReceivedRPSI)
+                {
+                    if ((info->codecSpecific.VP8.pictureIdRPSI & 0x3fff) == (_pictureIDLastSentRef & 0x3fff)) // compare 14 bits
+                    {
+                        // remote peer have received our last reference frame
+                        // switch frame type
+                        _haveReceivedAcknowledgement = true;
+                        _nextRefIsGolden = !_nextRefIsGolden;
+                        _pictureIDLastAcknowledgedRef = _pictureIDLastSentRef;
+                    }
+                }
+            }
+            const WebRtc_UWord16 periodX = 64; // we need a period X to decide on the distance between golden and altref
+            if (_pictureID % periodX == 0)
+            {
+                // only required if we have had a loss
+                // however we don't acknowledge a SLI so if that is lost it's no good
+                flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame
+
+                if (_nextRefIsGolden)
+                {
+                    flags |= VP8_EFLAG_FORCE_GF; // force a golden
+                    flags |= VP8_EFLAG_NO_UPD_ARF; // don't update altref
+                    if (_haveReceivedAcknowledgement)
+                    {
+                        // we can't set this if we refer to a key frame
+                        // pw temporary as proof of concept
+                        flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame
+                    }
+                }
+                else
+                {
+                    flags |= VP8_EFLAG_FORCE_ARF; // force an altref
+                    flags |= VP8_EFLAG_NO_UPD_GF; // Don't update golden
+                    if (_haveReceivedAcknowledgement)
+                    {
+                        // we can't set this if we refer to a key frame
+                        // pw temporary as proof of concept
+                        flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alternate reference frame
+                    }
+                }
+                // remember our last reference frame
+                _pictureIDLastSentRef = _pictureID;
+            }
+            else
+            {
+                flags |= VP8_EFLAG_NO_UPD_GF;  // don't update golden
+                flags |= VP8_EFLAG_NO_UPD_ARF; // don't update altref
+            }
+        }
+#endif
+        _encodedImage._frameType = kDeltaFrame;
+    }
+
+    if (vpx_codec_encode(_encoder, _raw, _maxFrameRate * inputImage._timeStamp / VP8_FREQ_HZ, 1, flags, VPX_DL_REALTIME))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    const vpx_codec_cx_pkt_t *pkt= vpx_codec_get_cx_data(_encoder, &iter); // no lagging => 1 frame at a time
+    if (pkt == NULL && !_encoder->err)
+    {
+        // dropped frame
+        return WEBRTC_VIDEO_CODEC_OK;
+    }
+    else if (pkt->kind == VPX_CODEC_CX_FRAME_PKT)
+    {
+        // attach Picture ID
+        // we use 14 bits generating 1 or 2 bytes
+        // TODO(hlundin): update to follow latest RTP spec
+        WebRtc_UWord8 pictureIdSize = 2;
+        // TODO(hlundin): we should refactor this so that the pictureID is
+        // signaled through a codec specific struct and added in the RTP module.
+        if (_pictureID > 0x7f)
+        {
+            // more than 7 bits
+            _encodedImage._buffer[0] = 0x80 | (WebRtc_UWord8)(_pictureID >> 7);
+            _encodedImage._buffer[1] = (WebRtc_UWord8)(_pictureID & 0x7f);
+        }
+        else
+        {
+            _encodedImage._buffer[0] = (WebRtc_UWord8)_pictureID;
+            pictureIdSize = 1;
+        }
+
+        memcpy(_encodedImage._buffer+pictureIdSize, pkt->data.frame.buf, pkt->data.frame.sz);
+        _encodedImage._length = WebRtc_UWord32(pkt->data.frame.sz) + pictureIdSize;
+        _encodedImage._encodedHeight = _raw->h;
+        _encodedImage._encodedWidth = _raw->w;
+
+        // check if encoded frame is a key frame
+        if (pkt->data.frame.flags & VPX_FRAME_IS_KEY)
+        {
+            _encodedImage._frameType = kKeyFrame;
+        }
+
+        if (_encodedImage._length > 0)
+        {
+            _encodedImage._timeStamp = inputImage._timeStamp;
+
+            // Figure out where partition boundaries are located.
+            RTPFragmentationHeader fragInfo;
+            fragInfo.VerifyAndAllocateFragmentationHeader(2); // two partitions: 1st and 2nd
+
+            // First partition
+            fragInfo.fragmentationOffset[0] = 0;
+            WebRtc_UWord8 *firstByte = &_encodedImage._buffer[pictureIdSize];
+            WebRtc_UWord32 tmpSize = (firstByte[2] << 16) | (firstByte[1] << 8)
+                | firstByte[0];
+            fragInfo.fragmentationLength[0] = (tmpSize >> 5) & 0x7FFFF;
+            // Let the PictureID belong to the first partition.
+            fragInfo.fragmentationLength[0] += pictureIdSize;
+            fragInfo.fragmentationPlType[0] = 0; // not known here
+            fragInfo.fragmentationTimeDiff[0] = 0;
+
+            // Second partition
+            fragInfo.fragmentationOffset[1] = fragInfo.fragmentationLength[0];
+            fragInfo.fragmentationLength[1] = _encodedImage._length -
+                fragInfo.fragmentationLength[0];
+            fragInfo.fragmentationPlType[1] = 0; // not known here
+            fragInfo.fragmentationTimeDiff[1] = 0;
+
+            _encodedCompleteCallback->Encoded(_encodedImage, NULL, &fragInfo);
+        }
+
+        _pictureID++; // prepare next
+        return WEBRTC_VIDEO_CODEC_OK;
+    }
+    return WEBRTC_VIDEO_CODEC_ERROR;
+}
+
+WebRtc_Word32
+VP8Encoder::SetPacketLoss(WebRtc_UWord32 packetLoss)
+{
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
+{
+    _encodedCompleteCallback = callback;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VP8Decoder::VP8Decoder():
+    _inited(false),
+    _feedbackModeOn(false),
+    _decoder(NULL)
+{
+}
+
+VP8Decoder::~VP8Decoder()
+{
+    _inited = true; // in order to do the actual release
+    Release();
+}
+
+WebRtc_Word32
+VP8Decoder::Reset()
+{
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    InitDecode(NULL, 1);
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Decoder::InitDecode(const VideoCodec* inst,
+                                 WebRtc_Word32 numberOfCores)
+{
+    vp8_postproc_cfg_t  ppcfg;
+    WebRtc_Word32 retVal = Release();
+    if (retVal < 0 )
+    {
+        return retVal;
+    }
+    if (_decoder == NULL)
+    {
+        _decoder = new vpx_dec_ctx_t;
+    }
+#ifdef DEV_PIC_LOSS
+    if(inst && inst->codecType == kVideoCodecVP8)
+    {
+        _feedbackModeOn = inst->codecSpecific.VP8.feedbackModeOn;
+    }
+#endif
+
+    vpx_codec_dec_cfg_t  cfg;
+    cfg.threads = numberOfCores;
+    cfg.h = cfg.w = 0; // set after decode
+
+    if(vpx_codec_dec_init(_decoder, vpx_codec_vp8_dx(), NULL, 0))
+    {
+        return WEBRTC_VIDEO_CODEC_MEMORY;
+    }
+    // config post-processing settings for decoder
+    ppcfg.post_proc_flag   = VP8_DEBLOCK;
+    ppcfg.deblocking_level = 5; //Strength of deblocking filter. Valid range:[0,16]
+    //ppcfg.NoiseLevel     = 1; //Noise intensity. Valid range: [0,7]
+    vpx_codec_control(_decoder, VP8_SET_POSTPROC, &ppcfg);
+
+    _inited = true;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Decoder::Decode(const EncodedImage& inputImage,
+                             bool missingFrames,
+                             const void* /*codecSpecificInfo*/,
+                             WebRtc_Word64 /*renderTimeMs*/)
+ {
+    if (!_inited)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (inputImage._buffer == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (_decodeCompleteCallback == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (inputImage._length <= 0)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    if (inputImage._completeFrame == false)
+    {
+        // future improvement
+        // we can't decode this frame
+        if (_feedbackModeOn)
+        {
+            return WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI;
+        }
+        else
+        {
+            return WEBRTC_VIDEO_CODEC_ERROR;
+        }
+    }
+    vpx_dec_iter_t _iter = NULL;
+    vpx_image_t* img;
+    WebRtc_UWord64 pictureID = 0;
+
+    // scan for number of bytes used for picture ID
+    WebRtc_UWord8 numberOfBytes;
+    for (numberOfBytes = 0;(inputImage._buffer[numberOfBytes] & 0x80 )&& numberOfBytes < 8; numberOfBytes++)
+    {
+        pictureID += inputImage._buffer[numberOfBytes] & 0x7f;
+        pictureID <<= 7;
+    }
+    pictureID += inputImage._buffer[numberOfBytes] & 0x7f;
+    numberOfBytes++;
+
+    // check for missing frames
+    if (missingFrames)
+    {
+        // call decoder with zero data length to signal missing frames
+        if (vpx_codec_decode(_decoder, NULL, 0, 0, VPX_DL_REALTIME))
+        {
+            return WEBRTC_VIDEO_CODEC_ERROR;
+        }
+    }
+
+    // we remove the picture ID here
+    if (vpx_codec_decode(_decoder,
+                         inputImage._buffer+numberOfBytes,
+                         inputImage._length-numberOfBytes,
+                         0,
+                         VPX_DL_REALTIME))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    int lastRefUpdates = 0;
+#ifdef DEV_PIC_LOSS
+    if (vpx_codec_control(_decoder, VP8D_GET_LAST_REF_UPDATES, &lastRefUpdates))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    int corrupted = 0;
+    if (vpx_codec_control(_decoder, VP8D_GET_FRAME_CORRUPTED, &corrupted))
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+#endif
+
+    img = vpx_codec_get_frame(_decoder, &_iter);
+
+    // Allocate memory for decoded image
+    WebRtc_UWord32 requiredSize = (3*img->h*img->w) >> 1;
+    if (_decodedImage._buffer != NULL)
+    {
+        delete [] _decodedImage._buffer;
+        _decodedImage._buffer = NULL;
+    }
+    if (_decodedImage._buffer == NULL)
+    {
+        _decodedImage._size = requiredSize;
+        _decodedImage._buffer = new WebRtc_UWord8[_decodedImage._size];
+        if (_decodedImage._buffer == NULL)
+        {
+            return WEBRTC_VIDEO_CODEC_MEMORY;
+        }
+    }
+
+    WebRtc_UWord8* buf;
+    WebRtc_UWord32 locCnt = 0;
+    WebRtc_UWord32 plane, y;
+
+    for (plane = 0; plane < 3; plane++)
+    {
+        buf = img->planes[plane];
+        WebRtc_UWord32 shiftFactor = plane ? 1 : 0;
+        for(y = 0; y < img->d_h >> shiftFactor; y++)
+        {
+            memcpy(&_decodedImage._buffer[locCnt], buf, img->d_w >> shiftFactor);
+            locCnt += img->d_w >> shiftFactor;
+            buf += img->stride[plane];
+        }
+    }
+
+    // Set image parameters
+    _decodedImage._height = img->d_h;
+    _decodedImage._width = img->d_w;
+    _decodedImage._length = (3 * img->d_h * img->d_w) >> 1;
+    _decodedImage._timeStamp = inputImage._timeStamp;
+    _decodeCompleteCallback->Decoded(_decodedImage);
+
+    // we need to communicate that we should send a RPSI with a specific picture ID
+
+    // TODO(pw): how do we know it's a golden or alt reference frame? On2 will provide an API
+    // for now I added it temporarily
+    if((lastRefUpdates & VP8_GOLD_FRAME) || (lastRefUpdates & VP8_ALTR_FRAME))
+    {
+        if (!missingFrames && (inputImage._completeFrame == true))
+        //if (!corrupted) // TODO(pw): Can we engage this line intead of the above?
+        {
+            _decodeCompleteCallback->ReceivedDecodedReferenceFrame(pictureID);
+        }
+    }
+    _decodeCompleteCallback->ReceivedDecodedFrame(pictureID);
+
+#ifdef DEV_PIC_LOSS
+    if (corrupted)
+    {
+        // we can decode but with artifacts
+        return WEBRTC_VIDEO_CODEC_REQUEST_SLI;
+    }
+#endif
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Decoder::RegisterDecodeCompleteCallback(DecodedImageCallback* callback)
+{
+    _decodeCompleteCallback = callback;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VP8Decoder::Release()
+{
+    if (_decodedImage._buffer != NULL)
+    {
+        delete [] _decodedImage._buffer;
+        _decodedImage._buffer = NULL;
+    }
+    if (_decoder != NULL)
+    {
+        if(vpx_codec_destroy(_decoder))
+        {
+            return WEBRTC_VIDEO_CODEC_MEMORY;
+        }
+        delete _decoder;
+        _decoder = NULL;
+    }
+
+    _inited = false;
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+}

modules/video_coding/codecs/vp8/main/source/vp8.gyp

@@ -0,0 +1,94 @@
+# Copyright (c) 2009 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+  'includes': [
+    '../../../../../../common_settings.gypi', # Common settings
+  ],
+  'targets': [
+    {
+      'target_name': 'webrtc_vp8',
+      'type': '<(library)',
+      'dependencies': [
+        '../../../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
+      ],
+      'include_dirs': [
+        '../interface',
+        '../../../interface',
+        '../../../../../interface',
+      ],
+      'conditions': [
+        ['build_with_chromium==1', {
+          'conditions': [
+            ['OS=="win"', {
+              'dependencies': [
+                # We don't want to link with the static library inside Chromium
+                # on Windows. Chromium uses the ffmpeg DLL and exports the
+                # necessary libvpx symbols for us.
+                '../../../../../../../libvpx/libvpx.gyp:libvpx_include',
+              ],
+            },{
+              'dependencies': [
+                '../../../../../../../libvpx/libvpx.gyp:libvpx',
+              ],
+              'include_dirs': [
+                '../../../../../../../libvpx/source/libvpx',
+              ],
+            }],
+          ],
+        },{
+          'dependencies': [
+            '../../../../../../../third_party/libvpx/libvpx.gyp:libvpx',
+          ],
+          'include_dirs': [
+            '../../../../../../../third_party/libvpx',
+          ],
+        }],
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '../interface',
+          '../../../interface',
+        ],
+      },
+      'sources': [
+        '../interface/vp8.h',
+        'vp8.cc',
+      ],
+    },
+
+    {
+      'target_name': 'vp8_test',
+      'type': 'executable',
+      'dependencies': [
+        'webrtc_vp8',
+        '../../../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '../../../test_framework/test_framework.gyp:test_framework',
+        '../../../../../../common_video/vplib/main/source/vplib.gyp:webrtc_vplib'
+      ],
+      'sources': [
+        # header files
+        '../test/benchmark.h',
+        '../test/normal_async_test.h',
+        '../test/packet_loss_test.h',
+        '../test/unit_test.h',
+        '../test/dual_decoder_test.h',
+
+        # source files
+        '../test/benchmark.cc',
+        '../test/normal_async_test.cc',
+        '../test/packet_loss_test.cc',
+        '../test/tester.cc',
+        '../test/unit_test.cc',
+        '../test/dual_decoder_test.cc',
+      ],
+    },
+  ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:

modules/video_coding/codecs/vp8/main/test/benchmark.cc

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "benchmark.h"
+#include "vp8.h"
+
+using namespace webrtc;
+
+VP8Benchmark::VP8Benchmark()
+:
+Benchmark("VP8Benchmark", "VP8 benchmark over a range of test cases", "../../VP8Benchmark.txt", "VP8")
+{
+}
+
+VP8Benchmark::VP8Benchmark(std::string name, std::string description)
+:
+Benchmark(name, description, "../../VP8Benchmark.txt", "VP8")
+{
+}
+
+VP8Benchmark::VP8Benchmark(std::string name, std::string description, std::string resultsFileName)
+:
+Benchmark(name, description, resultsFileName, "VP8")
+{
+}
+
+VideoEncoder*
+VP8Benchmark::GetNewEncoder()
+{
+    return new VP8Encoder();
+}
+
+VideoDecoder*
+VP8Benchmark::GetNewDecoder()
+{
+    return new VP8Decoder();
+}

modules/video_coding/codecs/vp8/main/test/benchmark.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
+
+#include "../../../test_framework/benchmark.h"
+
+class VP8Benchmark : public Benchmark
+{
+public:
+    VP8Benchmark();
+    VP8Benchmark(std::string name, std::string description);
+    VP8Benchmark(std::string name, std::string description, std::string resultsFileName);
+
+protected:
+    virtual webrtc::VideoEncoder* GetNewEncoder();
+    virtual webrtc::VideoDecoder* GetNewDecoder();
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_

modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc

@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "dual_decoder_test.h"
+
+#include <assert.h>
+#include <string.h> // memcmp
+#include <time.h>
+
+VP8DualDecoderTest::VP8DualDecoderTest(float bitRate)
+:
+VP8NormalAsyncTest(bitRate)
+{
+    _decoder2 = NULL;
+}
+
+VP8DualDecoderTest::VP8DualDecoderTest() 
+:
+_decoder2(NULL),
+VP8NormalAsyncTest("VP8 Dual Decoder Test", "Tests VP8 dual decoder", 1)
+{}
+
+VP8DualDecoderTest::~VP8DualDecoderTest()
+{
+    if(_decoder2)
+    {
+        _decoder2->Release();
+        delete _decoder2;
+    }
+
+    _decodedVideoBuffer2.Free();
+}
+
+void
+VP8DualDecoderTest::Perform()
+{
+    _inname = "test/testFiles/foreman_cif.yuv";
+    CodecSettings(352, 288, 30, _bitRate);
+    Setup();
+    _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
+    _decodedVideoBuffer2.VerifyAndAllocate(_lengthSourceFrame);
+    if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
+    {
+        exit(EXIT_FAILURE);
+    }
+    _decoder->InitDecode(&_inst,1);
+
+    FrameQueue frameQueue;
+    VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
+    DualDecoderCompleteCallback decCallback(&_decodedVideoBuffer);
+    DualDecoderCompleteCallback decCallback2(&_decodedVideoBuffer2);
+    _encoder->RegisterEncodeCompleteCallback(&encCallback);
+    _decoder->RegisterDecodeCompleteCallback(&decCallback);
+    if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
+    {
+        exit(EXIT_FAILURE);
+    }
+    _totalEncodeTime = _totalDecodeTime = 0;
+    _totalEncodePipeTime = _totalDecodePipeTime = 0;
+    bool complete = false;
+    _framecnt = 0;
+    _encFrameCnt = 0;
+    _decFrameCnt = 0;
+    _sumEncBytes = 0;
+    _lengthEncFrame = 0;
+    double starttime = clock()/(double)CLOCKS_PER_SEC;
+    while (!complete)
+    {
+        if (_encFrameCnt == 10)
+        {
+            // initialize second decoder and copy state
+            _decoder2 = static_cast<webrtc::VP8Decoder *>(_decoder->Copy());
+            assert(_decoder2 != NULL);
+            _decoder2->RegisterDecodeCompleteCallback(&decCallback2);
+        }
+        CodecSpecific_InitBitrate();
+        complete = Encode();
+        if (!frameQueue.Empty() || complete)
+        {
+            while (!frameQueue.Empty())
+            {   
+                _frameToDecode =
+                    static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
+                int lost = DoPacketLoss();
+                if (lost == 2)
+                {
+                    // Lost the whole frame, continue
+                    _missingFrames = true;
+                    delete _frameToDecode;
+                    _frameToDecode = NULL;
+                    continue;
+                }
+                int ret = Decode(lost);
+                delete _frameToDecode;
+                _frameToDecode = NULL;
+                if (ret < 0)
+                {
+                    fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
+                    exit(EXIT_FAILURE);
+                }
+                else if (ret == 0)
+                {
+                    _framecnt++;
+                }
+                else
+                {
+                    fprintf(stderr,
+                        "\n\nPositive return value from decode!\n\n");
+                }
+            }
+        }
+    }
+    double endtime = clock()/(double)CLOCKS_PER_SEC;
+    double totalExecutionTime = endtime - starttime;
+    printf("Total execution time: %.1f s\n", totalExecutionTime);
+    _sumEncBytes = encCallback.EncodedBytes();
+    double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
+    double avgEncTime = _totalEncodeTime / _encFrameCnt;
+    double avgDecTime = _totalDecodeTime / _decFrameCnt;
+    printf("Actual bitrate: %f kbps\n", actualBitRate);
+    printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
+    printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
+    printf("Average encode pipeline time: %.1f ms\n",
+           1000 * _totalEncodePipeTime / _encFrameCnt);
+    printf("Average decode pipeline  time: %.1f ms\n",
+           1000 * _totalDecodePipeTime / _decFrameCnt);
+    printf("Number of encoded frames: %u\n", _encFrameCnt);
+    printf("Number of decoded frames: %u\n", _decFrameCnt);
+    (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
+        _bitRate << " kbps" << std::endl;
+    (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
+    (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
+    _encoder->Release();
+    _decoder->Release();
+    Teardown();
+}
+
+
+int
+VP8DualDecoderTest::Decode(int lossValue)
+{
+    _sumEncBytes += _frameToDecode->_frame->GetLength();
+    double starttime = 0;
+    webrtc::EncodedImage encodedImage;
+    VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
+    encodedImage._completeFrame = !lossValue;
+    _decodeCompleteTime = 0;
+    _decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
+    int ret = _decoder->Decode(encodedImage, _missingFrames,
+        _frameToDecode->_codecSpecificInfo);
+    // second decoder
+    if (_decoder2)
+    {
+        int ret2 = _decoder2->Decode(encodedImage, _missingFrames,
+            _frameToDecode->_codecSpecificInfo, 0 /* dummy */);
+
+        // check return values
+        if (ret < 0 || ret2 < 0 || ret2 != ret)
+        {
+            exit(EXIT_FAILURE);
+        }
+
+        // compare decoded images
+        if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
+            _decodedVideoBuffer.GetLength(),
+            _decodedVideoBuffer2.GetBuffer(), _decodedVideoBuffer.GetLength()))
+        {
+            fprintf(stderr,"\n\nClone output different from master.\n\n");
+            exit(EXIT_FAILURE);
+        }
+
+    }
+
+    _missingFrames = false;
+    return ret;
+}
+
+
+bool
+VP8DualDecoderTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes, 
+                                    const void* ptrB, unsigned int bLengthBytes)
+{
+    if (aLengthBytes != bLengthBytes)
+    {
+        return false;
+    }
+
+    return memcmp(ptrA, ptrB, aLengthBytes) == 0;
+}
+
+WebRtc_Word32 DualDecoderCompleteCallback::Decoded(webrtc::RawImage& image)
+{
+    _decodedVideoBuffer->VerifyAndAllocate(image._length);
+    _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
+    _decodedVideoBuffer->SetWidth(image._width);
+    _decodedVideoBuffer->SetHeight(image._height);
+    _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
+    _decodeComplete = true;
+    return 0;
+}
+
+bool DualDecoderCompleteCallback::DecodeComplete()
+{
+    if (_decodeComplete)
+    {
+        _decodeComplete = false;
+        return true;
+    }
+    return false;
+}
+

modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
+
+#include "vp8.h"
+#include "normal_async_test.h"
+
+class DualDecoderCompleteCallback;
+
+class VP8DualDecoderTest : public VP8NormalAsyncTest
+{
+public:
+    VP8DualDecoderTest(float bitRate);
+    VP8DualDecoderTest();
+    virtual ~VP8DualDecoderTest();
+    virtual void Perform();
+protected:
+    VP8DualDecoderTest(std::string name, std::string description,
+                       unsigned int testNo)
+    : VP8NormalAsyncTest(name, description, testNo) {}
+    virtual int Decode(int lossValue = 0);
+
+    webrtc::VP8Decoder*     _decoder2;
+    TestVideoBuffer         _decodedVideoBuffer2;
+    static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes, 
+        const void *ptrB, unsigned int bLengthBytes);
+private:
+};
+
+class DualDecoderCompleteCallback : public webrtc::DecodedImageCallback
+{
+public:
+    DualDecoderCompleteCallback(TestVideoBuffer* buffer)
+    : _decodedVideoBuffer(buffer), _decodeComplete(false) {}
+    WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
+    bool DecodeComplete();
+private:
+    TestVideoBuffer* _decodedVideoBuffer;
+    bool _decodeComplete;
+};
+
+
+#endif

modules/video_coding/codecs/vp8/main/test/normal_async_test.cc

@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "normal_async_test.h"
+
+using namespace webrtc;
+
+VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate) :
+    NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, 1),
+    _hasReceivedRPSI(false)
+{
+}
+
+VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo):
+    NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, testNo),
+    _hasReceivedRPSI(false)
+{
+}
+
+void
+VP8NormalAsyncTest::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
+{
+    if (bitRate > 0)
+    {
+        _bitRate = bitRate;
+
+    }else if (_bitRate == 0)
+    {
+        _bitRate = 600;
+    }
+    _inst.codecType = kVideoCodecVP8;
+    _inst.codecSpecific.VP8.feedbackModeOn = true;
+    _inst.codecSpecific.VP8.pictureLossIndicationOn = true;
+    _inst.codecSpecific.VP8.complexity;
+    _inst.maxFramerate = (unsigned char)frameRate;
+    _inst.startBitrate = _bitRate;
+    _inst.maxBitrate = 8000;
+    _inst.width = width;
+    _inst.height = height;
+}
+
+void
+VP8NormalAsyncTest::CodecSpecific_InitBitrate()
+{
+    if (_bitRate == 0)
+    {
+        _encoder->SetRates(600, _inst.maxFramerate);
+    }else
+    {
+         _encoder->SetRates(_bitRate, _inst.maxFramerate);
+    }
+}
+
+WebRtc_Word32
+VP8NormalAsyncTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
+{
+    _pictureIdRPSI = pictureId;
+    _hasReceivedRPSI = true;
+    return 0;
+}
+
+void*
+VP8NormalAsyncTest::CreateEncoderSpecificInfo() const
+{
+    CodecSpecificInfo* vp8CodecSpecificInfo = new CodecSpecificInfo();
+    vp8CodecSpecificInfo->codecType = kVideoCodecVP8;
+    vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
+    vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
+    vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = false;
+
+    _hasReceivedRPSI = false;
+
+    return vp8CodecSpecificInfo;
+}

modules/video_coding/codecs/vp8/main/test/normal_async_test.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
+
+#include "../../../test_framework/normal_async_test.h"
+
+class VP8NormalAsyncTest : public NormalAsyncTest
+{
+public:
+    VP8NormalAsyncTest(WebRtc_UWord32 bitRate);
+    VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo);
+    VP8NormalAsyncTest() : NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", 1) {}
+protected:
+    VP8NormalAsyncTest(std::string name, std::string description, unsigned int testNo) : NormalAsyncTest(name, description, testNo) {}
+    virtual void CodecSpecific_InitBitrate();
+    virtual void CodecSettings(int width, int height, WebRtc_UWord32 frameRate=30, WebRtc_UWord32 bitRate=0);
+    virtual void* CreateEncoderSpecificInfo() const;
+    virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+private:
+    mutable bool  _hasReceivedRPSI;
+    WebRtc_UWord64  _pictureIdRPSI;
+};
+
+#endif

modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet_loss_test.h"
+#include <cassert>
+
+VP8PacketLossTest::VP8PacketLossTest()
+:
+PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode")
+{
+}
+
+VP8PacketLossTest::VP8PacketLossTest(std::string name, std::string description)
+:
+PacketLossTest(name, description)
+{
+}
+
+VP8PacketLossTest::VP8PacketLossTest(double lossRate, bool useNack)
+:
+PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode", lossRate, useNack)
+{
+}
+
+void
+VP8PacketLossTest::CodecSpecific_InitBitrate()
+{
+    assert(_bitRate > 0);
+    WebRtc_UWord32 simulatedBitRate;
+    if (_lossProbability != _lossRate)
+    {
+        // Simulating NACK
+        simulatedBitRate = (WebRtc_UWord32)(_bitRate / (1 + _lossRate));
+    }
+    else
+    {
+        simulatedBitRate = _bitRate;
+    }
+    _encoder->SetRates(simulatedBitRate, _inst.maxFramerate);
+}
+
+int VP8PacketLossTest::ByteLoss(int size, unsigned char* /* pkg */, int bytesToLose)
+{
+    int retLength = size - bytesToLose;
+    if (retLength < 4)
+    {
+        retLength = 4;
+    }
+    return retLength;
+}

modules/video_coding/codecs/vp8/main/test/packet_loss_test.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
+
+#include "../../../test_framework/packet_loss_test.h"
+
+class VP8PacketLossTest : public PacketLossTest
+{
+public:
+    VP8PacketLossTest();
+    VP8PacketLossTest(double lossRate, bool useNack);
+
+protected:
+    VP8PacketLossTest(std::string name, std::string description);
+    virtual void CodecSpecific_InitBitrate();
+    virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
+
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_

modules/video_coding/codecs/vp8/main/test/tester.cc

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet_loss_test.h"
+#include "benchmark.h"
+#include "unit_test.h"
+#include "normal_async_test.h"
+#include "dual_decoder_test.h"
+#include "vp8.h"
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+using namespace webrtc;
+
+void PopulateTests(std::vector<Test*>* tests)
+{
+    tests->push_back(new VP8UnitTest());
+//    tests->push_back(new VP8DualDecoderTest());
+//    tests->push_back(new VP8Benchmark());
+//    tests->push_back(new VP8PacketLossTest());
+//    tests->push_back(new VP8NormalAsyncTest());
+}
+
+int main()
+{
+    VP8Encoder* enc;
+    VP8Decoder* dec;
+    std::vector<Test*> tests;
+    PopulateTests(&tests);
+    std::fstream log;
+    log.open("../../TestLog.txt", std::fstream::out | std::fstream::app);
+    std::vector<Test*>::iterator it;
+    for (it = tests.begin() ; it < tests.end(); it++)
+    {
+        enc = new VP8Encoder();
+        dec = new VP8Decoder();
+        (*it)->SetEncoder(enc);
+        (*it)->SetDecoder(dec);
+        (*it)->SetLog(&log);
+        (*it)->Perform();
+        (*it)->Print();
+        delete enc;
+        delete dec;
+        delete *it;
+    }
+   log.close();
+   tests.pop_back();
+   return 0;
+}

modules/video_coding/codecs/vp8/main/test/unit_test.cc

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "unit_test.h"
+#include "../../../test_framework/video_source.h"
+#include "vp8.h"
+
+#include <string.h>
+
+using namespace webrtc;
+
+VP8UnitTest::VP8UnitTest()
+:
+UnitTest("VP8UnitTest", "Unit test")
+{
+}
+
+VP8UnitTest::VP8UnitTest(std::string name, std::string description)
+:
+UnitTest(name, description)
+{
+}
+
+void
+VP8UnitTest::Print()
+{
+    WebRtc_Word8 versionStr[64];
+
+    // GetVersion tests.
+
+      VIDEO_TEST(_encoder->Version(versionStr, sizeof(versionStr)) > 0);
+//    printf("\n%s", versionStr);
+//    UnitTest::Print();
+}
+
+WebRtc_UWord32
+VP8UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate, WebRtc_UWord32 /*frameRate*/)
+{
+    bitRate = _encoder->SetRates(bitRate, _inst.maxFramerate);
+    VIDEO_TEST_EXIT_ON_ERR(bitRate >= 0);
+    return bitRate;
+}
+
+bool
+VP8UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
+                             const void* ptrB, unsigned int bLengthBytes)
+{
+    const unsigned char* cPtrA = (const unsigned char*)ptrA;
+    const unsigned char* cPtrB = (const unsigned char*)ptrB;
+    // Skip picture ID before comparing
+    int aSkip = PicIdLength(cPtrA);
+    int bSkip = PicIdLength(cPtrB);
+    return UnitTest::CheckIfBitExact(cPtrA + aSkip, aLengthBytes,
+                                     cPtrB + bSkip, bLengthBytes);
+}
+
+int
+VP8UnitTest::PicIdLength(const unsigned char* ptr)
+{
+    WebRtc_UWord8 numberOfBytes;
+    WebRtc_UWord64 pictureID = 0;
+    for (numberOfBytes = 0; (ptr[numberOfBytes] & 0x80) && numberOfBytes < 8; numberOfBytes++)
+    {
+        pictureID += ptr[numberOfBytes] & 0x7f;
+        pictureID <<= 7;
+    }
+    pictureID += ptr[numberOfBytes] & 0x7f;
+    numberOfBytes++;
+    return numberOfBytes;
+}
+
+void
+VP8UnitTest::Perform()
+{
+    Setup();
+    FILE *outFile = NULL;
+    std::string outFileName;
+    VP8Encoder* enc = (VP8Encoder*)_encoder;
+    VP8Decoder* dec = (VP8Decoder*)_decoder;
+    int frameLength = 0;
+
+    //----- Encoder parameter tests -----
+    //-- Calls before InitEncode() --
+    VIDEO_TEST(enc->Release() == WEBRTC_VIDEO_CODEC_OK);
+    VIDEO_TEST(enc->SetRates(_bitRate, _inst.maxFramerate) == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+    VIDEO_TEST(enc->SetRates(_bitRate, _inst.maxFramerate) == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+   // VIDEO_TEST(enc->GetCodecConfigParameters(configParameters, sizeof(configParameters)) ==
+   //     WEBRTC_VIDEO_CODEC_UNINITIALIZED);
+
+
+    VideoCodec codecInst;
+    strncpy(codecInst.plName, "VP8", 31);
+    codecInst.plType = 126;
+    codecInst.maxBitrate = 0;
+    codecInst.minBitrate = 0;
+    codecInst.width = 1440;
+    codecInst.height = 1080;
+    codecInst.maxFramerate = 30;
+    codecInst.startBitrate = 300;
+    codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+    VIDEO_TEST(enc->InitEncode(&codecInst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+
+
+    //-- Test two problematic level settings --
+    strncpy(codecInst.plName, "VP8", 31);
+    codecInst.plType = 126;
+    codecInst.maxBitrate = 0;
+    codecInst.minBitrate = 0;
+    codecInst.width = 352;
+    codecInst.height = 288;
+    codecInst.maxFramerate = 30;
+    codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+    codecInst.startBitrate = 300;
+    VIDEO_TEST(enc->InitEncode(&codecInst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+
+    // Settings not correct for this profile
+    strncpy(codecInst.plName, "VP8", 31);
+    codecInst.plType = 126;
+    codecInst.maxBitrate = 0;
+    codecInst.minBitrate = 0;
+    codecInst.width = 176;
+    codecInst.height = 144;
+    codecInst.maxFramerate = 15;
+    codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
+    codecInst.startBitrate = 300;
+    //VIDEO_TEST(enc->InitEncode(&codecInst, 1, 1440) == WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED);
+
+    VIDEO_TEST_EXIT_ON_ERR(enc->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
+
+
+    //-- ProcessNewBitrate() errors --
+    // Bad bitrate.
+    VIDEO_TEST(enc->SetRates(_inst.maxBitrate + 1, _inst.maxFramerate) == WEBRTC_VIDEO_CODEC_OK);
+
+   // Signaling not used.
+
+    // Bad packetloss.
+//    VIDEO_TEST(enc->SetPacketLoss(300) < 0);
+
+    //----- Decoder parameter tests -----
+    //-- Calls before InitDecode() --
+    VIDEO_TEST(dec->Release() == 0);
+    VIDEO_TEST_EXIT_ON_ERR(dec->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
+
+    //-- SetCodecConfigParameters() errors --
+    unsigned char tmpBuf[128];
+    VIDEO_TEST(dec->SetCodecConfigParameters(NULL, sizeof(tmpBuf)) == -1);
+    VIDEO_TEST(dec->SetCodecConfigParameters(tmpBuf, 1) == -1);
+   // Garbage data.
+    VIDEO_TEST(dec->SetCodecConfigParameters(tmpBuf, sizeof(tmpBuf)) == -1);
+
+    //----- Function tests -----
+    outFileName = "../../" + _source->GetName() + "-errResTest.yuv";
+    outFile = fopen(outFileName.c_str(), "wb");
+    VIDEO_TEST_EXIT_ON_ERR(outFile != NULL);
+
+    UnitTest::Perform();
+    Teardown();
+
+}

modules/video_coding/codecs/vp8/main/test/unit_test.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
+
+#include "../../../test_framework/unit_test.h"
+
+class VP8UnitTest : public UnitTest
+{
+public:
+    VP8UnitTest();
+    VP8UnitTest(std::string name, std::string description);
+    virtual void Perform();
+    virtual void Print();
+
+protected:
+    virtual WebRtc_UWord32 CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
+                                                    WebRtc_UWord32 /*frameRate*/);
+    virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
+                                 const void *ptrB, unsigned int bLengthBytes);
+    static int PicIdLength(const unsigned char* ptr);
+};
+
+////////////////////////////////////////////////////////////////
+// RESERVATIONS TO PASSING UNIT TEST ON VP8 CODEC             //
+// Test that will not pass:                                   //
+// 1. Check bit exact for decoded images.                     //
+// 2. Target bitrate - Allow a margin of 10% instead of 5%    //
+// 3. Detecting errors in bit stream - NA.                    //
+////////////////////////////////////////////////////////////////
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_

modules/video_coding/main/OWNERS

@@ -0,0 +1,4 @@
+holmer@google.com
+mikhal@google.com
+marpan@google.com
+hlundin@google.com

modules/video_coding/main/interface/video_coding.h

@@ -0,0 +1,495 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
+#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
+
+#include "module.h"
+#include "module_common_types.h"
+#include "video_coding_defines.h"
+
+namespace webrtc
+{
+
+class VideoEncoder;
+class VideoDecoder;
+
+class VideoCodingModule : public Module
+{
+public:
+    static VideoCodingModule* Create(const WebRtc_Word32 id);
+
+    static void Destroy(VideoCodingModule* module);
+
+    // Get number of supported codecs
+    //
+    // Return value     : Number of supported codecs
+    static WebRtc_UWord8 NumberOfCodecs();
+
+    // Get supported codec settings with using id
+    //
+    // Input:
+    //      - listId         : Id or index of the codec to look up
+    //      - codec          : Memory where the codec settings will be stored
+    //
+    // Return value     : VCM_OK,              on success
+    //                    VCM_PARAMETER_ERROR  if codec not supported or id too high
+    static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, VideoCodec* codec);
+
+    // Get supported codec settings using codec type
+    //
+    // Input:
+    //      - codecType      : The codec type to get settings for
+    //      - codec          : Memory where the codec settings will be stored
+    //
+    // Return value     : VCM_OK,              on success
+    //                    VCM_PARAMETER_ERROR  if codec not supported
+    static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* codec);
+
+    /*
+    *   Sender
+    */
+
+    // Any encoder-related state of VCM will be initialized to the
+    // same state as when the VCM was created. This will not interrupt
+    // or effect decoding functionality of VCM. VCM will lose all the
+    // encoding-related settings by calling this function.
+    // For instance, a send codec has to be registered again.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 InitializeSender() = 0;
+
+    // Resets the encoder state to the same state as when the encoder
+    // was created.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 ResetEncoder() = 0;
+
+    // Registers a codec to be used for encoding. Calling this
+    // API multiple times overwrites any previously registered codecs.
+    //
+    // Input:
+    //      - sendCodec      : Settings for the codec to be registered.
+    //      - numberOfCores  : The number of cores the codec is allowed
+    //                         to use.
+    //      - maxPayloadSize : The maximum size each payload is allowed
+    //                                to have. Usually MTU - overhead.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
+                                            WebRtc_UWord32 numberOfCores,
+                                            WebRtc_UWord32 maxPayloadSize) = 0;
+
+    // API to get the current send codec in use.
+    //
+    // Input:
+    //      - currentSendCodec : Address where the sendCodec will be written.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const = 0;
+
+    // API to get the current send codec type
+    //
+    // Return value      : Codec type, on success.
+    //                     kVideoCodecUnknown, on error or if no send codec is set
+    virtual VideoCodecType SendCodec() const = 0;
+
+    // Register an external encoder object. This can not be used together with
+    // external decoder callbacks.
+    //
+    // Input:
+    //      - externalEncoder : Encoder object to be used for encoding frames inserted
+    //                          with the AddVideoFrame API.
+    //      - payloadType     : The payload type bound which this encoder is bound to.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                                  WebRtc_UWord8 payloadType,
+                                                  bool internalSource = false) = 0;
+
+    // API to get codec config parameters to be sent out-of-band to a receiver.
+    //
+    // Input:
+    //      - buffer          : Memory where the codec config parameters should be written.
+    //      - size            : Size of the memory available.
+    //
+    // Return value      : Number of bytes written, on success.
+    //                     < 0,                     on error.
+    virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size) = 0;
+
+    // API to get currently configured encoder target bit rate.
+    //
+    // Return value      : The encoder target bit rate, on success.
+    //                     < 0,                         on error.
+    virtual WebRtc_UWord32 Bitrate() const = 0;
+
+    // API to get currently configured encoder target frame rate.
+    //
+    // Return value      : The encoder target frame rate, on success.
+    //                     < 0,                           on error.
+    virtual WebRtc_UWord32 FrameRate() const = 0;
+
+    // Sets the parameters describing the send channel. These parameters are inputs to the
+    // Media Optimization inside the VCM and also specifies the target bit rate for the
+    // encoder. Bit rate used by NACK should already be compensated for by the user.
+    //
+    // Input:
+    //      - availableBandWidth    : Band width available for the VCM in kbit/s.
+    //      - lossRate              : Fractions of lost packets the past second.
+    //                                (loss rate in percent = 100 * packetLoss / 255)
+    //      - RTT                   : Current round-trip time in ms.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 availableBandWidth,
+                                               WebRtc_UWord8 lossRate,
+                                               WebRtc_UWord32 RTT) = 0;
+
+    // Sets the parameters describing the receive channel. These parameters are inputs to the
+    // Media Optimization inside the VCM.
+    //
+    // Input:
+    //      - RTT                   : Current round-trip time in ms.
+    //                                with the most amount available bandwidth in a conference
+    //                                scenario
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 RTT) = 0;
+
+    // Register a transport callback which will be called to deliver the encoded data and
+    // side information.
+    //
+    // Input:
+    //      - transport  : The callback object to register.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
+
+    // Register video output information callback which will be called to deliver information
+    // about the video stream produced by the encoder, for instance the average frame rate and
+    // bit rate.
+    //
+    // Input:
+    //      - outputInformation  : The callback object to register.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterSendStatisticsCallback(
+                                     VCMSendStatisticsCallback* sendStats) = 0;
+
+    // Register a video quality settings callback which will be called when
+    // frame rate/dimensions need to be updated for video quality optimization
+    //
+    // Input:
+    //		- videoQMSettings  : The callback object to register.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error
+    virtual WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings) = 0;
+
+    // Register a video protection callback which will be called to deliver
+    // the requested FEC rate and NACK status (on/off).
+    //
+    // Input:
+    //      - protection  : The callback object to register.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
+
+    // Enable or disable a video protection method.
+    //
+    // Input:
+    //      - videoProtection  : The method to enable or disable.
+    //      - enable           : True if the method should be enabled, false if
+    //                           it should be disabled.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
+                                             bool enable) = 0;
+
+    // Add one raw video frame to the encoder. This function does all the necessary
+    // processing, then decides what frame type to encode, or if the frame should be
+    // dropped. If the frame should be encoded it passes the frame to the encoder
+    // before it returns.
+    //
+    // Input:
+    //      - videoFrame        : Video frame to encode.
+    //      - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 AddVideoFrame(const VideoFrame& videoFrame,
+                                        const VideoContentMetrics* _contentMetrics = NULL,
+                                        const void* codecSpecificInfo = NULL) = 0;
+
+    // Next frame encoded should be of the type frameType.
+    //
+    // Input:
+    //      - frameType    : The frame type to encode next time a VideoFrame
+    //                       is added to the module.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 FrameTypeRequest(FrameType frameType) = 0;
+
+    // Frame Dropper enable. Can be used to disable the frame dropping when the encoder
+    // over-uses its bit rate. This API is designed to be used when the encoded frames
+    // are supposed to be stored to an AVI file, or when the I420 codec is used and the
+    // target bit rate shouldn't affect the frame rate.
+    //
+    // Input:
+    //      - enable            : True to enable the setting, false to disable it.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 EnableFrameDropper(bool enable) = 0;
+
+    // Sent frame counters
+    virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const = 0;
+
+    /*
+    *   Receiver
+    */
+
+    // The receiver state of the VCM will be initialized to the
+    // same state as when the VCM was created. This will not interrupt
+    // or effect the send side functionality of VCM. VCM will lose all the
+    // decoding-related settings by calling this function. All frames
+    // inside the jitter buffer are flushed and the delay is reset.
+    // For instance, a receive codec has to be registered again.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 InitializeReceiver() = 0;
+
+    // Register possible receive codecs, can be called multiple times for different codecs.
+    // The module will automatically switch between registered codecs depending on the
+    // payload type of incoming frames. The actual decoder will be created when needed.
+    //
+    // Input:
+    //      - receiveCodec      : Settings for the codec to be registered.
+    //      - numberOfCores     : Number of CPU cores that the decoder is allowed to use.
+    //      - requireKeyFrame   : Set this to true if you don't want any delta frames
+    //                            to be decoded until the first key frame has been decoded.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                                               WebRtc_Word32 numberOfCores,
+                                               bool requireKeyFrame = false) = 0;
+
+    // Register an externally defined decoder/renderer object. Can be a decoder only or a
+    // decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
+    // be used for decoding incoming streams.
+    //
+    // Input:
+    //      - externalDecoder        : The external decoder/renderer object.
+    //      - payloadType            : The payload type which this decoder should be
+    //                                 registered to.
+    //      - internalRenderTiming   : True if the internal renderer (if any) of the decoder
+    //                                 object can make sure to render at a given time in ms.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                                  WebRtc_UWord8 payloadType,
+                                                  bool internalRenderTiming) = 0;
+
+    // Register a receive callback. Will be called whenever there is a new frame ready
+    // for rendering.
+    //
+    // Input:
+    //      - receiveCallback        : The callback object to be used by the module when a
+    //                                 frame is ready for rendering.
+    //                                 De-register with a NULL pointer.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
+
+    // Register a receive statistics callback which will be called to deliver information
+    // about the video stream received by the receiving side of the VCM, for instance the
+    // average frame rate and bit rate.
+    //
+    // Input:
+    //      - receiveStats  : The callback object to register.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
+                               VCMReceiveStatisticsCallback* receiveStats) = 0;
+
+    // Register a frame type request callback. This callback will be called when the
+    // module needs to request specific frame types from the send side.
+    //
+    // Input:
+    //      - frameTypeCallback      : The callback object to be used by the module when
+    //                                 requesting a specific type of frame from the send side.
+    //                                 De-register with a NULL pointer.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 RegisterFrameTypeCallback(
+                                  VCMFrameTypeCallback* frameTypeCallback) = 0;
+
+    // Register a frame storage callback. This callback will be called right before an
+    // encoded frame is given to the decoder. Useful for recording the incoming video sequence.
+    //
+    // Input:
+    //      - frameStorageCallback    : The callback object used by the module
+    //                                  to store a received encoded frame.
+    //
+    // Return value     : VCM_OK, on success.
+    //                    < 0,         on error.
+    virtual WebRtc_Word32 RegisterFrameStorageCallback(
+                             VCMFrameStorageCallback* frameStorageCallback) = 0;
+
+    // Registers a callback which is called whenever the receive side of the VCM
+    // encounters holes in the packet sequence and needs packets to be retransmitted.
+    //
+    // Input:
+    //              - callback      : The callback to be registered in the VCM.
+    //
+    // Return value     : VCM_OK,     on success.
+    //                    <0,              on error.
+    virtual WebRtc_Word32 RegisterPacketRequestCallback(
+                                        VCMPacketRequestCallback* callback) = 0;
+
+    // Waits for the next frame in the jitter buffer to become complete
+    // (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
+    // Should be called as often as possible to get the most out of the decoder.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
+
+    // Waits for the next frame in the dual jitter buffer to become complete
+    // (waits no longer than maxWaitTimeMs), then passes it to the dual decoder
+    // for decoding. This will never trigger a render callback. Should be
+    // called frequently, and as long as it returns 1 it should be called again
+    // as soon as possible.
+    //
+    // Return value      : 1,           if a frame was decoded
+    //                     0,           if no frame was decoded
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
+
+    // Decodes a frame and sets an appropriate render time in ms relative to the system time.
+    // Should be used in conjunction with VCMFrameStorageCallback.
+    //
+    // Input:
+    //      - frameFromStorage      : Encoded frame read from file or received through
+    //                                the VCMFrameStorageCallback callback.
+    //
+    // Return value:        : VCM_OK, on success
+    //                        < 0,         on error
+    virtual WebRtc_Word32 DecodeFromStorage(const EncodedVideoData& frameFromStorage) = 0;
+
+    // Reset the decoder state to the initial state.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 ResetDecoder() = 0;
+
+    // API to get the codec which is currently used for decoding by the module.
+    //
+    // Input:
+    //      - currentReceiveCodec      : Settings for the codec to be registered.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
+
+    // API to get the codec type currently used for decoding by the module.
+    //
+    // Return value      : codecy type,            on success.
+    //                     kVideoCodecUnknown, on error or if no receive codec is registered
+    virtual VideoCodecType ReceiveCodec() const = 0;
+
+    // Insert a parsed packet into the receiver side of the module. Will be placed in the
+    // jitter buffer waiting for the frame to become complete. Returns as soon as the packet
+    // has been placed in the jitter buffer.
+    //
+    // Input:
+    //      - incomingPayload      : Payload of the packet.
+    //      - payloadLength        : Length of the payload.
+    //      - rtpInfo              : The parsed header.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
+                                       WebRtc_UWord32 payloadLength,
+                                       const WebRtcRTPHeader& rtpInfo) = 0;
+
+    // Sets codec config parameters received out-of-band to the currently
+    // selected receive codec.
+    //
+    // Input:
+    //      - payloadType   : Payload type which specifies which codec to set these
+    //                        parameters to.
+    //      - buffer        : Codec config parameters.
+    //      - length        : Length of the parameter data.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetCodecConfigParameters(WebRtc_UWord8 payloadType,
+                                                   const WebRtc_UWord8* buffer,
+                                                   WebRtc_Word32 length) = 0;
+
+    // Minimum playout delay (Used for lip-sync). This is the minimum delay required
+    // to sync with audio. Not included in  VideoCodingModule::Delay()
+    // Defaults to 0 ms.
+    //
+    // Input:
+    //      - minPlayoutDelayMs   : Additional delay in ms.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs) = 0;
+
+    // Set the time required by the renderer to render a frame.
+    //
+    // Input:
+    //      - timeMS        : The time in ms required by the renderer to render a frame.
+    //
+    // Return value      : VCM_OK, on success.
+    //                     < 0,         on error.
+    virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS) = 0;
+
+    // The total delay desired by the VCM. Can be less than the minimum
+    // delay set with SetMinimumPlayoutDelay.
+    //
+    // Return value      : Total delay in ms, on success.
+    //                     < 0,               on error.
+    virtual WebRtc_Word32 Delay() const = 0;
+
+    // Get the received frame counters. Keeps track of the number of each frame type
+    // received since the start of the call.
+    //
+    // Output:
+    //      - frameCount      : Struct to be filled with the number of frames received.
+    //
+    // Return value           : VCM_OK,        on success.
+    //                          <0,                 on error.
+    virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_

modules/video_coding/main/interface/video_coding_defines.h

@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+
+// Error codes
+#define VCM_REQUEST_SLI          2
+#define VCM_MISSING_CALLBACK     1
+#define VCM_OK                   0
+#define VCM_GENERAL_ERROR       -1
+#define VCM_LEVEL_EXCEEDED      -2
+#define VCM_MEMORY              -3
+#define VCM_PARAMETER_ERROR     -4
+#define VCM_UNKNOWN_PAYLOAD     -5
+#define VCM_CODEC_ERROR         -6
+#define VCM_UNINITIALIZED       -7
+#define VCM_NO_CODEC_REGISTERED -8
+#define VCM_JITTER_BUFFER_ERROR -9
+#define VCM_OLD_PACKET_ERROR    -10
+#define VCM_NO_FRAME_DECODED    -11
+#define VCM_ERROR_REQUEST_SLI   -12
+#define VCM_NOT_IMPLEMENTED     -20
+
+#define VCM_H263_PAYLOAD_TYPE       34
+#define VCM_RED_PAYLOAD_TYPE        96
+#define VCM_ULPFEC_PAYLOAD_TYPE     97
+#define VCM_H263_1998_PAYLOAD_TYPE 121
+#define VCM_VP8_PAYLOAD_TYPE       120
+#define VCM_I420_PAYLOAD_TYPE      124
+
+enum VCMNackProperties
+{
+    kNackHistoryLength = 450
+};
+
+enum VCMH263FrameDrop
+{
+    kDecodePFrames,
+    kDropPFrames
+};
+
+enum VCMVideoProtection
+{
+    kProtectionNack,                // Both send-side and receive-side
+    kProtectionNackSender,          // Send-side only
+    kProtectionNackReceiver,        // Receive-side only
+    kProtectionDualDecoder,
+    kProtectionFEC,
+    kProtectionNackFEC,
+    kProtectionKeyOnLoss,
+    kProtectionKeyOnKeyLoss,
+    kProtectionPeriodicKeyFrames
+};
+
+enum VCMTemporalDecimation
+{
+    kBitrateOverUseDecimation,
+};
+
+struct VCMFrameCount
+{
+    WebRtc_UWord32 numKeyFrames;
+    WebRtc_UWord32 numDeltaFrames;
+};
+
+
+// Callback class used for sending data ready to be packetized
+class VCMPacketizationCallback
+{
+public:
+    virtual WebRtc_Word32 SendData(const FrameType frameType,
+                                   const WebRtc_UWord8 payloadType,
+                                   const WebRtc_UWord32 timeStamp,
+                                   const WebRtc_UWord8* payloadData,
+                                   const WebRtc_UWord32 payloadSize,
+                                   const RTPFragmentationHeader& fragmentationHeader) = 0;
+protected:
+    virtual ~VCMPacketizationCallback() {}
+};
+
+// Callback class used for passing decoded frames which are ready to be rendered.
+class VCMFrameStorageCallback
+{
+public:
+    virtual WebRtc_Word32 StoreReceivedFrame(const EncodedVideoData& frameToStore) = 0;
+
+protected:
+    virtual ~VCMFrameStorageCallback() {}
+};
+
+// Callback class used for passing decoded frames which are ready to be rendered.
+class VCMReceiveCallback
+{
+public:
+    virtual WebRtc_Word32 FrameToRender(VideoFrame& videoFrame) = 0;
+    virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
+
+protected:
+    virtual ~VCMReceiveCallback() {}
+};
+
+// Callback class used for informing the user of the bit rate and frame rate produced by the
+// encoder.
+class VCMSendStatisticsCallback
+{
+public:
+    virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
+                                         const WebRtc_UWord32 frameRate) = 0;
+
+protected:
+    virtual ~VCMSendStatisticsCallback() {}
+};
+
+// Callback class used for informing the user of the incoming bit rate and frame rate.
+class VCMReceiveStatisticsCallback
+{
+public:
+    virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bitRate,
+                                            const WebRtc_UWord32 frameRate) = 0;
+
+protected:
+    virtual ~VCMReceiveStatisticsCallback() {}
+};
+
+// Callback class used for telling the user about the requested amount of bit stream protection
+// Key frame FEC rate, delta frame and whether NACK should be on or off.
+class VCMProtectionCallback
+{
+public:
+    virtual WebRtc_Word32 ProtectionRequest(const WebRtc_UWord8 deltaFECRate,
+                                            const WebRtc_UWord8 keyFECRate,
+                                            const bool nack) = 0;
+
+protected:
+    virtual ~VCMProtectionCallback() {}
+};
+
+// Callback class used for telling the user about what frame type needed to continue decoding.
+// Typically a key frame when the stream has been corrupted in some way.
+class VCMFrameTypeCallback
+{
+public:
+    virtual WebRtc_Word32 FrameTypeRequest(const FrameType frameType) = 0;
+    virtual WebRtc_Word32 SliceLossIndicationRequest(const WebRtc_UWord64 pictureId) {return -1;}
+
+protected:
+    virtual ~VCMFrameTypeCallback() {}
+};
+
+// Callback class used for telling the user about which packet sequence numbers are currently
+// missing and need to be resent.
+class VCMPacketRequestCallback
+{
+public:
+    virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
+                                        WebRtc_UWord16 length) = 0;
+
+protected:
+    virtual ~VCMPacketRequestCallback() {}
+};
+
+// Callback used to inform the user of the the desired resolution
+// as subscribed by Media Optimization (Quality Modes)
+class VCMQMSettingsCallback
+{
+public:
+    virtual WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
+                                             const WebRtc_UWord32 width,
+                                             const WebRtc_UWord32 height) = 0;
+
+protected:
+    virtual ~VCMQMSettingsCallback() {}
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_

modules/video_coding/main/source/codec_database.cc

@@ -0,0 +1,799 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "codec_database.h"
+#include "../../../../engine_configurations.h"
+#include "internal_defines.h"
+#include "trace.h"
+
+#if defined(_WIN32)
+    // VS 2005: Don't warn for default initialized arrays. See help for more info.
+    // Don't warn for strncpy being unsecure.
+    // switch statement contains 'default' but no 'case' labels
+#pragma warning(disable:4351; disable:4996; disable:4065)
+#endif
+
+// Supported codecs
+#ifdef  VIDEOCODEC_VP8
+    #include "vp8.h"
+#endif
+#ifdef  VIDEOCODEC_I420
+    #include "i420.h"
+#endif
+
+namespace webrtc
+{
+
+VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
+                                     WebRtc_UWord32 numberOfCores,
+                                     bool requireKeyFrame)
+:
+_settings(settings),
+_numberOfCores(numberOfCores),
+_requireKeyFrame(requireKeyFrame)
+{
+}
+
+VCMExtDecoderMapItem::VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
+                                           WebRtc_UWord8 payloadType,
+                                           bool internalRenderTiming)
+:
+_payloadType(payloadType),
+_externalDecoderInstance(externalDecoderInstance),
+_internalRenderTiming(internalRenderTiming)
+{
+}
+
+VCMCodecDataBase::VCMCodecDataBase(WebRtc_Word32 id):
+_id(id),
+_numberOfCores(0),
+_maxPayloadSize(kDefaultPayloadSize),
+_periodicKeyFrames(false),
+_currentEncIsExternal(false),
+_sendCodec(),
+_receiveCodec(),
+_externalPayloadType(0),
+_externalEncoder(NULL),
+_internalSource(false),
+_ptrEncoder(NULL),
+_ptrDecoder(NULL),
+_currentDecIsExternal(false),
+_decMap(),
+_decExternalMap()
+{
+    //
+}
+
+VCMCodecDataBase::~VCMCodecDataBase()
+{
+    Reset();
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Version(WebRtc_Word8* version,
+                                WebRtc_UWord32& remainingBufferInBytes,
+                                WebRtc_UWord32& position) const
+{
+    VCMGenericEncoder* encoder = NULL;
+    VideoCodec settings;
+    WebRtc_Word32 ret;
+    for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++)
+    {
+        ret = VCMCodecDataBase::Codec(i, &settings);
+        if (ret < 0)
+        {
+            return ret;
+        }
+        encoder = CreateEncoder(settings.codecType);
+        if (encoder == NULL)
+        {
+            return VCM_MEMORY;
+        }
+        ret = encoder->_encoder.Version(&version[position], remainingBufferInBytes);
+        if (ret < 0)
+        {
+            return ret;
+        }
+        remainingBufferInBytes -= ret;
+        position += ret;
+        delete &encoder->_encoder;
+        delete encoder;
+    }
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Reset()
+{
+    WebRtc_Word32 ret = ResetReceiver();
+    if (ret < 0)
+    {
+        return ret;
+    }
+    ret = ResetSender();
+    if (ret < 0)
+    {
+        return ret;
+    }
+   return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ResetSender()
+{
+    DeleteEncoder();
+    _periodicKeyFrames = false;
+    return VCM_OK;
+}
+
+VCMGenericEncoder*
+VCMCodecDataBase::CreateEncoder(VideoCodecType type) const
+{
+    switch(type)
+    {
+#ifdef  VIDEOCODEC_VP8
+        case kVideoCodecVP8:
+            return new VCMGenericEncoder(*(new VP8Encoder));
+        break;
+#endif
+#ifdef VIDEOCODEC_I420
+        case kVideoCodecI420:
+            return new VCMGenericEncoder(*(new I420Encoder));
+            break;
+#endif
+        default:
+        return NULL;
+        break;
+    }
+}
+
+void
+VCMCodecDataBase::DeleteEncoder()
+{
+    if (_ptrEncoder)
+    {
+        _ptrEncoder->Release();
+        if (!_currentEncIsExternal)
+        {
+            delete &_ptrEncoder->_encoder;
+        }
+        delete _ptrEncoder;
+        _ptrEncoder = NULL;
+    }
+}
+
+WebRtc_UWord8
+VCMCodecDataBase::NumberOfCodecs()
+{
+    return VCM_NUM_VIDEO_CODECS_AVAILABLE;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Codec(WebRtc_UWord8 listId, VideoCodec *settings)
+{
+    if (settings == NULL)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+
+    if (listId >= VCM_NUM_VIDEO_CODECS_AVAILABLE)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    memset(settings, 0, sizeof(VideoCodec));
+    switch (listId)
+    {
+#ifdef VIDEOCODEC_VP8
+    case VCM_VP8_IDX:
+        {
+            strncpy(settings->plName, "VP8", 3);
+            settings->codecType = kVideoCodecVP8;
+            // 96 to 127 dynamic payload types for video codecs
+            settings->plType = VCM_VP8_PAYLOAD_TYPE;
+            settings->startBitrate = 100;
+            settings->minBitrate = VCM_MIN_BITRATE;
+            settings->maxBitrate = 0;
+            settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
+            settings->width = VCM_DEFAULT_CODEC_WIDTH;
+            settings->height = VCM_DEFAULT_CODEC_HEIGHT;
+            break;
+        }
+#endif
+#ifdef VIDEOCODEC_I420
+    case VCM_I420_IDX:
+        {
+            strncpy(settings->plName, "I420", 4);
+            settings->codecType = kVideoCodecI420;
+            // 96 to 127 dynamic payload types for video codecs
+            settings->plType = VCM_I420_PAYLOAD_TYPE;
+            // Bitrate needed for this size and framerate
+            settings->startBitrate = 3*VCM_DEFAULT_CODEC_WIDTH*
+                                       VCM_DEFAULT_CODEC_HEIGHT*8*
+                                       VCM_DEFAULT_FRAME_RATE/1000/2;
+            settings->maxBitrate = settings->startBitrate;
+            settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
+            settings->width = VCM_DEFAULT_CODEC_WIDTH;
+            settings->height = VCM_DEFAULT_CODEC_HEIGHT;
+            settings->minBitrate = VCM_MIN_BITRATE;
+            break;
+        }
+#endif
+    default:
+        {
+            return VCM_PARAMETER_ERROR;
+        }
+    }
+
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::Codec(VideoCodecType codecType, VideoCodec* settings)
+{
+    for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++)
+    {
+        const WebRtc_Word32 ret = VCMCodecDataBase::Codec(i, settings);
+        if (ret != VCM_OK)
+        {
+            return ret;
+        }
+        if (codecType == settings->codecType)
+        {
+            return VCM_OK;
+        }
+    }
+    return VCM_PARAMETER_ERROR;
+}
+
+// assuming only one registered encoder - since only one used, no need for more
+WebRtc_Word32
+VCMCodecDataBase::RegisterSendCodec(const VideoCodec* sendCodec,
+                                    WebRtc_UWord32 numberOfCores,
+                                    WebRtc_UWord32 maxPayloadSize)
+ {
+    if (sendCodec == NULL)
+    {
+        return VCM_UNINITIALIZED;
+    }
+    if (maxPayloadSize == 0)
+    {
+        maxPayloadSize = kDefaultPayloadSize;
+    }
+    if (numberOfCores > 32)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    if (strcmp(sendCodec->plName, "H263") == 0 &&
+        (sendCodec->plType != 34))
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    if (sendCodec->plType <= 0)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    // Make sure the start bit rate is sane...
+    if (sendCodec->startBitrate > 1000000)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    if (sendCodec->codecType == kVideoCodecUnknown)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    _numberOfCores = numberOfCores;
+    _maxPayloadSize = maxPayloadSize;
+
+    memcpy(&_sendCodec, sendCodec, sizeof(VideoCodec));
+
+    if (_sendCodec.maxBitrate == 0)
+    {
+        // max is one bit per pixel
+        _sendCodec.maxBitrate = ((WebRtc_Word32)_sendCodec.height *
+                                 (WebRtc_Word32)_sendCodec.width *
+                                 (WebRtc_Word32)_sendCodec.maxFramerate) / 1000;
+        if (_sendCodec.startBitrate > _sendCodec.maxBitrate)
+        {
+            // but if the customer tries to set a higher start bit rate we will increase
+            // the max accordingly
+            _sendCodec.maxBitrate = _sendCodec.startBitrate;
+        }
+    }
+
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::SendCodec(VideoCodec* currentSendCodec) const
+{
+    WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id), "SendCodec");
+
+    if(_ptrEncoder == NULL)
+    {
+        return VCM_UNINITIALIZED;
+    }
+    memcpy(currentSendCodec, &_sendCodec, sizeof(VideoCodec));
+    return VCM_OK;
+}
+
+VideoCodecType
+VCMCodecDataBase::SendCodec() const
+{
+    WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id),
+            "SendCodec type");
+    if (_ptrEncoder == NULL)
+    {
+        return kVideoCodecUnknown;
+    }
+    return _sendCodec.codecType;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec)
+{
+    wasSendCodec = false;
+    if (_externalPayloadType != payloadType)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    if (_sendCodec.plType == payloadType)
+    {
+        //De-register as send codec if needed
+        DeleteEncoder();
+        memset(&_sendCodec, 0, sizeof(VideoCodec));
+        _currentEncIsExternal = false;
+        wasSendCodec = true;
+    }
+    _externalPayloadType = 0;
+    _externalEncoder = NULL;
+    _internalSource = false;
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                          WebRtc_UWord8 payloadType,
+                                          bool internalSource)
+{
+    // since only one encoder can be used at a given time,
+    // only one external encoder can be registered/used
+    _externalEncoder = externalEncoder;
+    _externalPayloadType = payloadType;
+    _internalSource = internalSource;
+
+    return VCM_OK;
+}
+
+VCMGenericEncoder*
+VCMCodecDataBase::SetEncoder(const VideoCodec* settings,
+                             VCMEncodedFrameCallback* VCMencodedFrameCallback)
+
+{
+    // if encoder exists, will destroy it and create new one
+    DeleteEncoder();
+
+    if (settings->plType == _externalPayloadType)
+    {
+        // External encoder
+        _ptrEncoder = new VCMGenericEncoder(*_externalEncoder, _internalSource);
+        _currentEncIsExternal = true;
+    }
+    else
+    {
+        _ptrEncoder = CreateEncoder(settings->codecType);
+        _currentEncIsExternal = false;
+    }
+
+    VCMencodedFrameCallback->SetPayloadType(settings->plType);
+
+    if (_ptrEncoder == NULL)
+    {
+        return NULL;
+    }
+
+    if (_ptrEncoder->InitEncode(settings, _numberOfCores, _maxPayloadSize) < 0)
+    {
+        DeleteEncoder();
+        return NULL;
+    }
+    else if (_ptrEncoder->RegisterEncodeCallback(VCMencodedFrameCallback) < 0)
+    {
+        DeleteEncoder();
+        return NULL;
+    }
+    // Intentionally don't check return value since the encoder registration
+    // shouldn't fail because the codec doesn't support changing the
+    // periodic key frame setting.
+    _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
+    return _ptrEncoder;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::SetPeriodicKeyFrames(bool enable)
+{
+    _periodicKeyFrames = enable;
+    if (_ptrEncoder != NULL)
+    {
+        return _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
+    }
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                                       WebRtc_UWord32 numberOfCores,
+                                       bool requireKeyFrame)
+{
+    WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCoding, VCMId(_id),
+                 "Codec: %s, Payload type %d, Height %d, Width %d, Bitrate %d, Framerate %d.",
+                 receiveCodec->plName, receiveCodec->plType,
+                 receiveCodec->height, receiveCodec->width,
+                 receiveCodec->startBitrate, receiveCodec->maxFramerate);
+
+    // check if payload value already exists, if so  - erase old and insert new
+    DeRegisterReceiveCodec(receiveCodec->plType);
+    if (receiveCodec->codecType == kVideoCodecUnknown)
+    {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    VideoCodec* newReceiveCodec = new VideoCodec(*receiveCodec);
+    _decMap.Insert(receiveCodec->plType,
+        new VCMDecoderMapItem(newReceiveCodec, numberOfCores, requireKeyFrame));
+
+    return VCM_OK;
+}
+
+WebRtc_Word32 VCMCodecDataBase::DeRegisterReceiveCodec(WebRtc_UWord8 payloadType)
+{
+    MapItem* item = _decMap.Find(payloadType);
+    if (item == NULL)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    VCMDecoderMapItem* decItem = static_cast<VCMDecoderMapItem*>(item->GetItem());
+    delete decItem->_settings;
+    delete decItem;
+    _decMap.Erase(item);
+    if (_receiveCodec.plType == payloadType)
+    {
+        // This codec is currently in use.
+        memset(&_receiveCodec, 0, sizeof(VideoCodec));
+        _currentDecIsExternal = false;
+    }
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ResetReceiver()
+{
+    ReleaseDecoder(_ptrDecoder);
+    _ptrDecoder = NULL;
+    memset(&_receiveCodec, 0, sizeof(VideoCodec));
+    MapItem* item = _decMap.First();
+    while (item != NULL)
+    {
+        VCMDecoderMapItem* decItem = static_cast<VCMDecoderMapItem*>(item->GetItem());
+        if (decItem != NULL)
+        {
+            if (decItem->_settings != NULL)
+            {
+                delete decItem->_settings;
+            }
+            delete decItem;
+        }
+        _decMap.Erase(item);
+        item = _decMap.First();
+    }
+    item = _decExternalMap.First();
+    while (item != NULL)
+    {
+        VCMExtDecoderMapItem* decItem = static_cast<VCMExtDecoderMapItem*>(item->GetItem());
+        if (decItem != NULL)
+        {
+            delete decItem;
+        }
+        _decExternalMap.Erase(item);
+        item = _decExternalMap.First();
+    }
+    _currentDecIsExternal = false;
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMCodecDataBase::DeRegisterExternalDecoder(WebRtc_UWord8 payloadType)
+{
+    MapItem* item = _decExternalMap.Find(payloadType);
+    if (item == NULL)
+    {
+        // Not found
+        return VCM_PARAMETER_ERROR;
+    }
+    if (_receiveCodec.plType == payloadType)
+    {
+        // Release it if it was registered and in use
+        ReleaseDecoder(_ptrDecoder);
+        _ptrDecoder = NULL;
+    }
+    DeRegisterReceiveCodec(payloadType);
+    VCMExtDecoderMapItem* decItem = static_cast<VCMExtDecoderMapItem*>(item->GetItem());
+    delete decItem;
+    _decExternalMap.Erase(item);
+    return VCM_OK;
+}
+
+// Add the external encoder object to the list of external decoders.
+// Won't be registered as a receive codec until RegisterReceiveCodec is called.
+WebRtc_Word32
+VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                          WebRtc_UWord8 payloadType,
+                                          bool internalRenderTiming)
+{
+    // check if payload value already exists, if so  - erase old and insert new
+    VCMExtDecoderMapItem* extDecoder = new VCMExtDecoderMapItem(externalDecoder,
+                                                                payloadType,
+                                                                internalRenderTiming);
+    if (extDecoder == NULL)
+    {
+        return VCM_MEMORY;
+    }
+    DeRegisterExternalDecoder(payloadType);
+    _decExternalMap.Insert(payloadType, extDecoder);
+
+    return VCM_OK;
+}
+
+bool
+VCMCodecDataBase::DecoderRegistered() const
+{
+    return (_decMap.Size() > 0);
+}
+
+WebRtc_Word32
+VCMCodecDataBase::ReceiveCodec(VideoCodec* currentReceiveCodec) const
+{
+    if (_ptrDecoder == NULL)
+    {
+        return VCM_NO_FRAME_DECODED;
+    }
+    memcpy(currentReceiveCodec, &_receiveCodec, sizeof(VideoCodec));
+    return VCM_OK;
+}
+
+VideoCodecType
+VCMCodecDataBase::ReceiveCodec() const
+{
+    if (_ptrDecoder == NULL)
+    {
+        return kVideoCodecUnknown;
+    }
+    return _receiveCodec.codecType;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::SetDecoder(WebRtc_UWord8 payloadType, VCMDecodedFrameCallback& callback)
+{
+    if (payloadType == _receiveCodec.plType || payloadType == 0)
+    {
+        return _ptrDecoder;
+    }
+    // check for exisitng decoder, if exists - delete
+    if (_ptrDecoder)
+    {
+        ReleaseDecoder(_ptrDecoder);
+        _ptrDecoder = NULL;
+        memset(&_receiveCodec, 0, sizeof(VideoCodec));
+    }
+    _ptrDecoder = CreateAndInitDecoder(payloadType, _receiveCodec, _currentDecIsExternal);
+    if (_ptrDecoder == NULL)
+    {
+        return NULL;
+    }
+    if (_ptrDecoder->RegisterDecodeCompleteCallback(&callback) < 0)
+    {
+        ReleaseDecoder(_ptrDecoder);
+        _ptrDecoder = NULL;
+        memset(&_receiveCodec, 0, sizeof(VideoCodec));
+        return NULL;
+    }
+    return _ptrDecoder;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateAndInitDecoder(WebRtc_UWord8 payloadType,
+                                       VideoCodec& newCodec,
+                                       bool &external) const
+{
+    VCMDecoderMapItem* decoderItem = FindDecoderItem(payloadType);
+    if (decoderItem == NULL)
+    {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id),
+                     "Unknown payload type: %u", payloadType);
+        return NULL;
+    }
+    VCMGenericDecoder* ptrDecoder = NULL;
+    VCMExtDecoderMapItem* externalDecItem = FindExternalDecoderItem(payloadType);
+    if (externalDecItem != NULL)
+    {
+        // External codec
+        ptrDecoder = new VCMGenericDecoder(*externalDecItem->_externalDecoderInstance, _id,
+                                           true);
+        external = true;
+    }
+    else
+    {
+        // create decoder
+        ptrDecoder = CreateDecoder(decoderItem->_settings->codecType);
+        external = false;
+    }
+    if (ptrDecoder == NULL)
+    {
+        return NULL;
+    }
+
+    if (ptrDecoder->InitDecode(decoderItem->_settings,
+                               decoderItem->_numberOfCores,
+                               decoderItem->_requireKeyFrame) < 0)
+    {
+        ReleaseDecoder(ptrDecoder);
+        return NULL;
+    }
+
+    SetCodecConfigParameters(*ptrDecoder, *decoderItem->_settings);
+
+    memcpy(&newCodec, decoderItem->_settings, sizeof(VideoCodec));
+    return ptrDecoder;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateDecoderCopy() const
+{
+    if (_ptrDecoder == NULL)
+    {
+        return NULL;
+    }
+    VideoDecoder* decoderCopy = _ptrDecoder->_decoder.Copy();
+    if (decoderCopy == NULL)
+    {
+        return NULL;
+    }
+    return new VCMGenericDecoder(*decoderCopy, _id, _ptrDecoder->External());
+}
+
+void
+VCMCodecDataBase::CopyDecoder(const VCMGenericDecoder& decoder)
+{
+    VideoDecoder* decoderCopy = decoder._decoder.Copy();
+    if (decoderCopy != NULL)
+    {
+        ReleaseDecoder(_ptrDecoder);
+        _ptrDecoder = new VCMGenericDecoder(*decoderCopy, _id, decoder.External());
+    }
+}
+
+bool
+VCMCodecDataBase::RenderTiming() const
+{
+    bool renderTiming = true;
+    if (_currentDecIsExternal)
+    {
+        VCMExtDecoderMapItem* extItem = FindExternalDecoderItem(_receiveCodec.plType);
+        renderTiming = extItem->_internalRenderTiming;
+    }
+    return renderTiming;
+}
+
+void
+VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const
+{
+    if (decoder != NULL)
+    {
+        decoder->Release();
+        if (!decoder->External() && &decoder->_decoder != NULL)
+        {
+            delete &decoder->_decoder;
+        }
+        delete decoder;
+    }
+}
+
+WebRtc_Word32
+VCMCodecDataBase::SetCodecConfigParameters(WebRtc_UWord8 payloadType,
+                                           const WebRtc_UWord8* buffer,
+                                           WebRtc_Word32 length)
+{
+    VCMDecoderMapItem* decItem = FindDecoderItem(payloadType);
+    if (decItem == NULL)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    switch (decItem->_settings->codecType)
+    {
+    case kVideoCodecMPEG4:
+    {
+        memcpy(decItem->_settings->codecSpecific.MPEG4.configParameters, buffer, length);
+        decItem->_settings->codecSpecific.MPEG4.configParametersSize =
+                static_cast<WebRtc_UWord8>(length);
+        break;
+    }
+    default:
+        // This codec doesn't have codec config parameters
+        return VCM_GENERAL_ERROR;
+    }
+    if (_ptrDecoder != NULL && _receiveCodec.plType == decItem->_settings->plType)
+    {
+        return _ptrDecoder->SetCodecConfigParameters(buffer, length);
+    }
+    return VCM_OK;
+}
+
+VCMDecoderMapItem*
+VCMCodecDataBase::FindDecoderItem(WebRtc_UWord8 payloadType) const
+{
+    MapItem* item = _decMap.Find(payloadType);
+    if (item != NULL)
+    {
+        return static_cast<VCMDecoderMapItem*>(item->GetItem());
+    }
+    return NULL;
+}
+
+VCMExtDecoderMapItem*
+VCMCodecDataBase::FindExternalDecoderItem(WebRtc_UWord8 payloadType) const
+{
+    MapItem* item = _decExternalMap.Find(payloadType);
+    if (item != NULL)
+    {
+        return static_cast<VCMExtDecoderMapItem*>(item->GetItem());
+    }
+    return NULL;
+}
+
+VCMGenericDecoder*
+VCMCodecDataBase::CreateDecoder(VideoCodecType type) const
+{
+    switch(type)
+    {
+#ifdef VIDEOCODEC_VP8
+    case kVideoCodecVP8:
+        return new VCMGenericDecoder(*(new VP8Decoder), _id);
+#endif
+#ifdef VIDEOCODEC_I420
+    case kVideoCodecI420:
+         return new VCMGenericDecoder(*(new I420Decoder), _id);
+#endif
+    default:
+        return NULL;
+    }
+}
+
+void
+VCMCodecDataBase::SetCodecConfigParameters(VCMGenericDecoder& decoder,
+                                           const VideoCodec& settings)
+{
+    switch (settings.codecType)
+    {
+    case kVideoCodecMPEG4:
+    {
+        if (settings.codecSpecific.MPEG4.configParametersSize > 0)
+        {
+            decoder.SetCodecConfigParameters(
+                                    settings.codecSpecific.MPEG4.configParameters,
+                                    settings.codecSpecific.MPEG4.configParametersSize);
+        }
+        break;
+    }
+    default:
+        // No codec config parameters for this codec
+        return;
+    }
+    return;
+}
+
+}

modules/video_coding/main/source/codec_database.h

@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+
+#include "video_coding.h"
+#include "video_codec_interface.h"
+#include "generic_decoder.h"
+#include "generic_encoder.h"
+#include "typedefs.h"
+#include "map_wrapper.h"
+
+namespace webrtc
+{
+
+enum VCMCodecDBProperties
+{
+    kDefaultPayloadSize = 1440
+};
+
+class VCMDecoderMapItem {
+public:
+    VCMDecoderMapItem(VideoCodec* settings,
+                      WebRtc_UWord32 numberOfCores,
+                      bool requireKeyFrame);
+
+    VideoCodec*     _settings;
+    WebRtc_UWord32  _numberOfCores;
+    bool            _requireKeyFrame;
+};
+
+class VCMExtDecoderMapItem {
+public:
+    VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
+                         WebRtc_UWord8 payloadType,
+                         bool internalRenderTiming);
+
+    WebRtc_UWord8   _payloadType;
+    VideoDecoder*   _externalDecoderInstance;
+    bool            _internalRenderTiming;
+};
+
+/*******************************/
+/* VCMCodecDataBase class      */
+/*******************************/
+class VCMCodecDataBase
+{
+public:
+    VCMCodecDataBase(WebRtc_Word32 id);
+    ~VCMCodecDataBase();
+    /**
+    * Fills "version" with the version of all codecs supported.
+    */
+    WebRtc_Word32 Version(WebRtc_Word8* version,
+                        WebRtc_UWord32& remainingBufferInBytes,
+                        WebRtc_UWord32& position) const;
+    /**
+    * Release codecdatabase - release all memory for both send and receive side
+    */
+    WebRtc_Word32 Reset();
+    /**
+    * Sender Side
+    */
+    /**
+    * Returns the number of supported codecs (or -1 in case of error).
+    */
+    static WebRtc_UWord8 NumberOfCodecs();
+    /**
+    * Get supported codecs with ID
+    * Input Values:
+    *       listnr    : Requested codec id number
+    *       codec_inst: Pointer to the struct in which the returned codec information is copied
+    * Return Values: 0 if successful, otherwise
+    */
+    static WebRtc_Word32 Codec(WebRtc_UWord8 listId, VideoCodec* settings);
+    static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* settings);
+    /**
+    * Reset Sender side
+    */
+    WebRtc_Word32 ResetSender();
+    /**
+    * Setting the sender side codec and initiaiting the desired codec given the VideoCodec
+    * struct.
+    * Return Value:	0 if the codec and the settings are supported, otherwise
+    */
+    WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
+                                  WebRtc_UWord32 numberOfCores,
+                                  WebRtc_UWord32 maxPayloadSize);
+    /**
+    * Get current send side codec. Relevant for internal codecs only.
+    */
+    WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const;
+    /**
+    * Get current send side codec type. Relevant for internal codecs only.
+    */
+    VideoCodecType SendCodec() const;
+    /**
+    * Register external encoder - current assumption - if one is registered then it will also
+    * be used, and therefore it is also initialized
+    * Return value: A pointer to the encoder on success, or null, in case of an error.
+    */
+    WebRtc_Word32 DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec);
+    WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                        WebRtc_UWord8 payloadType,
+                                        bool internalSource);
+    /**
+    * Returns a encoder given a payloadname - to be used with internal encoders only.
+    * Special cases:
+    *	 Encoder exists -  If payload matches, returns existing one, otherwise,
+    *	 deletes existing one and creates new one.
+    *	 No match found / Error - returns NULL.
+    */
+    VCMGenericEncoder* SetEncoder(const VideoCodec* settings,
+                                  VCMEncodedFrameCallback* VCMencodedFrameCallback);
+
+    WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
+
+    bool InternalSource() const;
+
+    /*
+    * Receiver Side
+    */
+    WebRtc_Word32 ResetReceiver();
+    /**
+    * Register external decoder/render object
+    */
+    WebRtc_Word32 DeRegisterExternalDecoder(WebRtc_UWord8 payloadType);
+    WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                        WebRtc_UWord8 payloadType,
+                                        bool internalRenderTiming);
+
+    bool DecoderRegistered() const;
+    /**
+    * Register recieve codec
+    */
+    WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                                     WebRtc_UWord32 numberOfCores,
+                                     bool requireKeyFrame);
+    WebRtc_Word32 DeRegisterReceiveCodec(WebRtc_UWord8 payloadType);
+    /**
+    * Get current receive side codec. Relevant for internal codecs only.
+    */
+    WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const;
+    /**
+    * Get current receive side codec type. Relevant for internal codecs only.
+    */
+    VideoCodecType ReceiveCodec() const;
+    /**
+    * Returns a decoder given which matches a payload type.
+    * Special cases:
+    *	 Decoder exists -  If payload matches, returns existing one, otherwise, deletes
+    *	 existing one, and creates new one.
+    *	 No match found / Error - returns NULL.
+    */
+    VCMGenericDecoder* SetDecoder(WebRtc_UWord8 payloadType, VCMDecodedFrameCallback& callback);
+
+    VCMGenericDecoder* CreateAndInitDecoder(WebRtc_UWord8 payloadType,
+                                            VideoCodec& newCodec,
+                                            bool &external) const;
+
+    VCMGenericDecoder* CreateDecoderCopy() const;
+
+    void ReleaseDecoder(VCMGenericDecoder* decoder) const;
+
+    void CopyDecoder(const VCMGenericDecoder& decoder);
+
+    bool RenderTiming() const;
+
+    WebRtc_Word32 SetCodecConfigParameters(WebRtc_UWord8 payloadType,
+                                         const WebRtc_UWord8* buffer,
+                                         WebRtc_Word32 length);
+
+protected:
+    /**
+    * Create an internal Encoder given a codec type
+    */
+    VCMGenericEncoder* CreateEncoder(VideoCodecType type) const;
+
+    void DeleteEncoder();
+    /*
+    * Create an internal Decoder given a codec type
+    */
+    VCMGenericDecoder* CreateDecoder(VideoCodecType type) const;
+
+    static void SetCodecConfigParameters(VCMGenericDecoder& decoder,
+                                         const VideoCodec& settings);
+
+    VCMDecoderMapItem* FindDecoderItem(WebRtc_UWord8 payloadType) const;
+
+    VCMExtDecoderMapItem* FindExternalDecoderItem(WebRtc_UWord8 payloadType) const;
+
+private:
+    WebRtc_Word32         _id;
+    WebRtc_UWord32        _numberOfCores;
+    WebRtc_UWord32        _maxPayloadSize;
+    bool                  _periodicKeyFrames;
+    bool                  _currentEncIsExternal;
+    VideoCodec            _sendCodec;
+    VideoCodec            _receiveCodec;
+    WebRtc_UWord8         _externalPayloadType;
+    VideoEncoder*         _externalEncoder;
+    bool                  _internalSource;
+    VCMGenericEncoder*    _ptrEncoder;
+    VCMGenericDecoder*    _ptrDecoder;
+    bool                  _currentDecIsExternal;
+    MapWrapper               _decMap;
+    MapWrapper               _decExternalMap;
+
+}; // end of VCMCodecDataBase class definition
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_

modules/video_coding/main/source/codec_timer.cc

@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "codec_timer.h"
+
+#include <assert.h>
+
+namespace webrtc
+{
+
+VCMCodecTimer::VCMCodecTimer()
+:
+_filteredMax(0),
+_firstDecodeTime(true),
+_shortMax(0),
+_history()
+{
+    Reset();
+}
+
+WebRtc_Word32 VCMCodecTimer::StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs)
+{
+    const WebRtc_Word32 timeDiff = static_cast<WebRtc_Word32>(nowMs - startTimeMs);
+    MaxFilter(timeDiff, nowMs);
+    return timeDiff;
+}
+
+void VCMCodecTimer::Reset()
+{
+    _filteredMax = 0;
+    _firstDecodeTime = true;
+    _shortMax = 0;
+    for (int i=0; i < MAX_HISTORY_SIZE; i++)
+    {
+        _history[i].shortMax = 0;
+        _history[i].timeMs = -1;
+    }
+}
+
+// Update the max-value filter
+void VCMCodecTimer::MaxFilter(WebRtc_Word32 decodeTime, WebRtc_Word64 nowMs)
+{
+    if (!_firstDecodeTime)
+    {
+        UpdateMaxHistory(decodeTime, nowMs);
+        ProcessHistory(nowMs);
+    }
+    else
+    {
+        _firstDecodeTime = false;
+    }
+}
+
+void
+VCMCodecTimer::UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now)
+{
+    if (_history[0].timeMs >= 0 &&
+        now - _history[0].timeMs < SHORT_FILTER_MS)
+    {
+        if (decodeTime > _shortMax)
+        {
+            _shortMax = decodeTime;
+        }
+    }
+    else
+    {
+        // Only add a new value to the history once a second
+        if(_history[0].timeMs == -1)
+        {
+            // First, no shift
+            _shortMax = decodeTime;
+        }
+        else
+        {
+            // Shift
+            for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
+            {
+                _history[i+1].shortMax = _history[i].shortMax;
+                _history[i+1].timeMs = _history[i].timeMs;
+            }
+        }
+        if (_shortMax == 0)
+        {
+            _shortMax = decodeTime;
+        }
+
+        _history[0].shortMax = _shortMax;
+        _history[0].timeMs = now;
+        _shortMax = 0;
+    }
+}
+
+void
+VCMCodecTimer::ProcessHistory(WebRtc_Word64 nowMs)
+{
+    _filteredMax = _shortMax;
+    if (_history[0].timeMs == -1)
+    {
+        return;
+    }
+    for (int i=0; i < MAX_HISTORY_SIZE; i++)
+    {
+        if (_history[i].timeMs == -1)
+        {
+            break;
+        }
+        if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
+        {
+            // This sample (and all samples after this) is too old
+            break;
+        }
+        if (_history[i].shortMax > _filteredMax)
+        {
+            // This sample is the largest one this far into the history
+            _filteredMax = _history[i].shortMax;
+        }
+    }
+}
+
+// Get the maximum observed time within a time window
+WebRtc_Word32 VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
+{
+    return _filteredMax;
+}
+
+}

modules/video_coding/main/source/codec_timer.h

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+
+// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
+#define MAX_HISTORY_SIZE 20
+#define SHORT_FILTER_MS 1000
+
+class VCMShortMaxSample
+{
+public:
+    VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
+
+    WebRtc_Word32     shortMax;
+    WebRtc_Word64     timeMs;
+};
+
+class VCMCodecTimer
+{
+public:
+    VCMCodecTimer();
+
+    // Updates and returns the max filtered decode time.
+    WebRtc_Word32 StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs);
+
+    // Empty the list of timers.
+    void Reset();
+
+    // Get the required decode time in ms.
+    WebRtc_Word32 RequiredDecodeTimeMs(FrameType frameType) const;
+
+private:
+    void UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now);
+    void MaxFilter(WebRtc_Word32 newTime, WebRtc_Word64 nowMs);
+    void ProcessHistory(WebRtc_Word64 nowMs);
+
+    WebRtc_Word32                     _filteredMax;
+    bool                              _firstDecodeTime;
+    WebRtc_Word32                     _shortMax;
+    VCMShortMaxSample                 _history[MAX_HISTORY_SIZE];
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_

modules/video_coding/main/source/content_metrics_processing.cc

@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "content_metrics_processing.h"
+#include "tick_time.h"
+#include "module_common_types.h"
+#include "video_coding_defines.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+//////////////////////////////////
+/// VCMContentMetricsProcessing //
+//////////////////////////////////
+
+VCMContentMetricsProcessing::VCMContentMetricsProcessing():
+_frameRate(0),
+_recAvgFactor(1 / 150.0f), // matched to  30fps
+_frameCnt(0),
+_prevAvgSizeZeroMotion(0),
+_avgSizeZeroMotion(0),
+_prevAvgSpatialPredErr(0),
+_avgSpatialPredErr(0),
+_frameCntForCC(0),
+_lastCCpdateTime(0)
+{
+    _globalRecursiveAvg = new VideoContentMetrics();
+}
+
+VCMContentMetricsProcessing::~VCMContentMetricsProcessing()
+{
+    delete _globalRecursiveAvg;
+}
+
+WebRtc_Word32
+VCMContentMetricsProcessing::Reset()
+{
+    _globalRecursiveAvg->Reset();
+    _frameCnt = 0;
+    _frameRate = 0;
+    //_recAvgFactor = 1 / 150.0f; // matched to 30 fps
+    _prevAvgSizeZeroMotion = 0;
+    _avgSizeZeroMotion = 0;
+    _prevAvgSpatialPredErr = 0;
+    _avgSpatialPredErr = 0;
+    _frameCntForCC = 0;
+
+    return VCM_OK;
+}
+void
+VCMContentMetricsProcessing::UpdateFrameRate(WebRtc_UWord32 frameRate)
+{
+    _frameRate = frameRate;
+    //Update recursive avg factor
+    _recAvgFactor = (float) 1000 / ((float)(_frameRate *  kQmMinIntervalMs));
+
+}
+
+WebRtc_Word32
+VCMContentMetricsProcessing::UpdateContentData(const VideoContentMetrics *contentMetrics)
+{
+    if (contentMetrics == NULL)
+    {
+        return VCM_OK;
+    }
+    return ProcessContent(contentMetrics);
+
+}
+
+VideoContentMetrics*
+VCMContentMetricsProcessing::Data()
+{
+    if (_frameCnt == 0)
+    {
+        return NULL;
+    }
+    return _globalRecursiveAvg;
+}
+
+WebRtc_UWord32
+VCMContentMetricsProcessing::ProcessContent(const VideoContentMetrics *contentMetrics)
+{
+    // update global metric
+    UpdateGlobalMetric(contentMetrics);
+
+    //Update metrics over local window for content change (CC) detection:
+    //two metrics are used for CC detection: size of zero motion, and spatial prediction error
+    //Not currently used:
+    //UpdateLocalMetricCC(contentMetrics->sizeZeroMotion, contentMetrics->spatialPredErr);
+
+    return VCM_OK;
+}
+
+bool
+VCMContentMetricsProcessing::ContentChangeCheck()
+{
+     bool result = false;
+
+    // Thresholds for bitrate and content change detection
+    float qmContentChangePercMotion = 0.4f;
+    float qmContentChangePercSpatial = 0.4f;
+
+    WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
+    if ( (now - _lastCCpdateTime) < kCcMinIntervalMs)
+    {
+        //keep averaging
+        return result;
+    }
+    else //check for detection and reset
+    {
+        //normalize
+        _avgSizeZeroMotion = _avgSizeZeroMotion / (float)(_frameCntForCC);
+        _prevAvgSpatialPredErr = _prevAvgSpatialPredErr / (float)(_frameCntForCC);
+
+        //check for content change
+        float diffMotion = fabs(_avgSizeZeroMotion - _prevAvgSizeZeroMotion);
+        float diffSpatial = fabs(_avgSpatialPredErr -_prevAvgSpatialPredErr);
+        if ((diffMotion > (_avgSizeZeroMotion * qmContentChangePercMotion)) ||
+            (diffSpatial > (_prevAvgSpatialPredErr * qmContentChangePercSpatial)))
+        {
+            result = true;
+        }
+
+        //copy to previous
+        _prevAvgSizeZeroMotion = _avgSizeZeroMotion;
+        _prevAvgSpatialPredErr = _avgSpatialPredErr;
+
+        //reset
+        _avgSizeZeroMotion = 0.;
+        _avgSpatialPredErr = 0.;
+        _frameCntForCC = 0;
+
+         _lastCCpdateTime = now;
+
+    }
+
+    return result;
+}
+
+//update metrics for content change detection: update is uniform average over soem time window
+void VCMContentMetricsProcessing::UpdateLocalMetricCC(float motionVal, float spatialVal)
+{
+
+    _frameCntForCC += 1;
+    _avgSizeZeroMotion += motionVal;
+    _avgSpatialPredErr += spatialVal;
+
+    return;
+
+}
+void
+VCMContentMetricsProcessing::UpdateGlobalMetric(const VideoContentMetrics *contentMetrics)
+{
+
+    // Threshold for size of zero motion cluster: for updating 3 metrics:
+    // motion magnitude, cluster distortion, and horizontalness
+    float nonZeroMvThr = 0.1f;
+
+    // first zero and one: take value as is (no motion search in frame zero).
+    float tmpRecAvgFactor  = _recAvgFactor;
+    if (_frameCnt < 1)
+    {
+        _recAvgFactor = 1;
+    }
+
+    _globalRecursiveAvg->motionPredErr = (1 - _recAvgFactor) * _globalRecursiveAvg->motionPredErr +
+                                          _recAvgFactor * contentMetrics->motionPredErr;
+
+    _globalRecursiveAvg->sizeZeroMotion = (1 - _recAvgFactor) * _globalRecursiveAvg->sizeZeroMotion +
+                                           _recAvgFactor * contentMetrics->sizeZeroMotion;
+
+    _globalRecursiveAvg->spatialPredErr = (1 - _recAvgFactor) * _globalRecursiveAvg->spatialPredErr +
+                                           _recAvgFactor * contentMetrics->spatialPredErr;
+
+    _globalRecursiveAvg->spatialPredErrH = (1 - _recAvgFactor) * _globalRecursiveAvg->spatialPredErrH +
+                                           _recAvgFactor * contentMetrics->spatialPredErrH;
+
+    _globalRecursiveAvg->spatialPredErrV = (1 - _recAvgFactor) * _globalRecursiveAvg->spatialPredErrV +
+                                           _recAvgFactor * contentMetrics->spatialPredErrV;
+
+    //motionMag metric is derived from NFD (normalized frame difference)
+    if (kNfdMetric == 1)
+    {
+        _globalRecursiveAvg->motionMagnitudeNZ = (1 - _recAvgFactor) * _globalRecursiveAvg->motionMagnitudeNZ +
+                                                    _recAvgFactor * contentMetrics->motionMagnitudeNZ;
+    }
+
+    if (contentMetrics->sizeZeroMotion > nonZeroMvThr)
+    {
+        _globalRecursiveAvg->motionClusterDistortion = (1 - _recAvgFactor) * _globalRecursiveAvg->motionClusterDistortion +
+                                                    _recAvgFactor *contentMetrics->motionClusterDistortion;
+
+        _globalRecursiveAvg->motionHorizontalness = (1 - _recAvgFactor) * _globalRecursiveAvg->motionHorizontalness +
+                                                _recAvgFactor * contentMetrics->motionHorizontalness;
+
+        //motionMag metric is derived from motion vectors
+        if (kNfdMetric == 0)
+        {
+            _globalRecursiveAvg->motionMagnitudeNZ = (1 - _recAvgFactor) * _globalRecursiveAvg->motionMagnitudeNZ +
+                                                _recAvgFactor * contentMetrics->motionMagnitudeNZ;
+        }
+    }
+
+    // update native values:
+    _globalRecursiveAvg->nativeHeight = contentMetrics->nativeHeight;
+    _globalRecursiveAvg->nativeWidth = contentMetrics->nativeWidth;
+    _globalRecursiveAvg->nativeFrameRate = contentMetrics->nativeFrameRate;
+
+    if (_frameCnt < 1)
+    {
+        _recAvgFactor = tmpRecAvgFactor;
+    }
+    _frameCnt++;
+    return;
+}
+
+}

modules/video_coding/main/source/content_metrics_processing.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+struct VideoContentMetrics;
+
+// QM interval time
+enum { kQmMinIntervalMs = 10000 };
+enum { kCcMinIntervalMs = 5000 };
+
+//Flag for NFD metric vs motion metric
+enum { kNfdMetric  = 1 };
+
+/**********************************/
+/* Content Metrics Processing     */
+/**********************************/
+class VCMContentMetricsProcessing
+{
+public:
+    VCMContentMetricsProcessing();
+    ~VCMContentMetricsProcessing();
+
+    // Update class with latest metrics
+    WebRtc_Word32 UpdateContentData(const VideoContentMetrics *contentMetrics);
+
+    // Check for content change detection
+    bool ContentChangeCheck();
+
+    //Initialize to
+    WebRtc_Word32 Reset();
+
+    // Inform class of current frame rate
+    void UpdateFrameRate(WebRtc_UWord32 frameRate);
+
+    // Get working (avg) value
+    VideoContentMetrics* Data();
+private:
+
+    // Compute working avg
+    WebRtc_UWord32 ProcessContent(const VideoContentMetrics *contentMetrics);
+
+    // Computation of global metric
+    void UpdateGlobalMetric(const VideoContentMetrics *contentMetrics);
+
+    // Compute local average of certain metrics for content change detection
+    void UpdateLocalMetricCC(float motionVal, float spatialVal);
+
+    VideoContentMetrics*    _globalRecursiveAvg;
+    WebRtc_UWord32          _frameRate;
+    float                   _recAvgFactor;
+    WebRtc_UWord32          _frameCnt;
+
+    float                   _prevAvgSizeZeroMotion;
+    float                   _avgSizeZeroMotion;
+    float                   _prevAvgSpatialPredErr;
+    float                   _avgSpatialPredErr;
+    WebRtc_UWord32          _frameCntForCC;
+    WebRtc_UWord64          _lastCCpdateTime;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_

modules/video_coding/main/source/encoded_frame.cc

@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "encoded_frame.h"
+#include "generic_encoder.h"
+#include "jitter_buffer_common.h"
+#include "video_coding_defines.h"
+
+namespace webrtc {
+
+VCMEncodedFrame::VCMEncodedFrame()
+:
+webrtc::EncodedImage(),
+_renderTimeMs(-1),
+_payloadType(0),
+_missingFrame(false),
+_codecSpecificInfo(NULL),
+_codecSpecificInfoLength(0),
+_codec(kVideoCodecUnknown)
+{
+}
+
+VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
+:
+webrtc::EncodedImage(rhs),
+_renderTimeMs(-1),
+_payloadType(0),
+_missingFrame(false),
+_codecSpecificInfo(NULL),
+_codecSpecificInfoLength(0),
+_codec(kVideoCodecUnknown)
+{
+    _buffer = NULL;
+    _size = NULL;
+    _length = NULL;
+    if (rhs._buffer != NULL)
+    {
+        VerifyAndAllocate(rhs._length);
+        memcpy(_buffer, rhs._buffer, rhs._length);
+    }
+}
+
+VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
+:
+webrtc::EncodedImage(rhs),
+_renderTimeMs(rhs._renderTimeMs),
+_payloadType(rhs._payloadType),
+_missingFrame(rhs._missingFrame),
+_codecSpecificInfo(NULL),
+_codecSpecificInfoLength(0),
+_codec(rhs._codec)
+{
+    _buffer = NULL;
+    _size = NULL;
+    _length = NULL;
+    if (rhs._buffer != NULL)
+    {
+        VerifyAndAllocate(rhs._size);
+        memcpy(_buffer, rhs._buffer, rhs._length);
+    }
+}
+
+VCMEncodedFrame::~VCMEncodedFrame()
+{
+    Free();
+}
+
+void VCMEncodedFrame::Free()
+{
+    Reset();
+    if (_buffer != NULL)
+    {
+        delete [] _buffer;
+        _buffer = NULL;
+    }
+}
+
+void VCMEncodedFrame::Reset()
+{
+    _renderTimeMs = -1;
+    _timeStamp = 0;
+    _payloadType = 0;
+    _codecSpecificInfo = NULL;
+    _codecSpecificInfoLength = 0;
+    _frameType = kDeltaFrame;
+    _encodedWidth = 0;
+    _encodedHeight = 0;
+    _completeFrame = false;
+    _missingFrame = false;
+    _length = 0;
+    _codec = kVideoCodecUnknown;
+}
+
+WebRtc_Word32
+VCMEncodedFrame::Store(VCMFrameStorageCallback& storeCallback) const
+{
+    EncodedVideoData frameToStore;
+    frameToStore.codec = _codec;
+    if (_buffer != NULL)
+    {
+        frameToStore.VerifyAndAllocate(_length);
+        memcpy(frameToStore.payloadData, _buffer, _length);
+        frameToStore.payloadSize = _length;
+    }
+    frameToStore.completeFrame = _completeFrame;
+    frameToStore.encodedWidth = _encodedWidth;
+    frameToStore.encodedHeight = _encodedHeight;
+    frameToStore.frameType = ConvertFrameType(_frameType);
+    frameToStore.missingFrame = _missingFrame;
+    frameToStore.payloadType = _payloadType;
+    frameToStore.renderTimeMs = _renderTimeMs;
+    frameToStore.timeStamp = _timeStamp;
+    storeCallback.StoreReceivedFrame(frameToStore);
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMEncodedFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
+{
+    if(minimumSize > _size)
+    {
+        // create buffer of sufficient size
+        WebRtc_UWord8* newBuffer = new WebRtc_UWord8[minimumSize];
+        if (newBuffer == NULL)
+        {
+            return -1;
+        }
+        if(_buffer)
+        {
+            // copy old data
+            memcpy(newBuffer, _buffer, _size);
+            delete [] _buffer;
+        }
+        _buffer = newBuffer;
+        _size = minimumSize;
+    }
+    return 0;
+}
+
+webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
+{
+    switch(frameType)
+    {
+    case kKeyFrame:
+        {
+            return  kVideoFrameKey;
+        }
+    case kDeltaFrame:
+        {
+            return kVideoFrameDelta;
+        }
+    case kGoldenFrame:
+        {
+            return kVideoFrameGolden;
+        }
+    case kAltRefFrame:
+        {
+            return kVideoFrameAltRef;
+        }
+    default:
+        {
+            return kVideoFrameDelta;
+        }
+    }
+}
+
+VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frameType)
+{
+    switch (frameType)
+    {
+    case kVideoFrameKey:
+        {
+            return kKeyFrame;
+        }
+    case kVideoFrameDelta:
+        {
+            return kDeltaFrame;
+        }
+    case kVideoFrameGolden:
+        {
+            return kGoldenFrame;
+        }
+    case kVideoFrameAltRef:
+        {
+            return kAltRefFrame;
+        }
+    default:
+        {
+            return kDeltaFrame;
+        }
+    }
+}
+
+}

modules/video_coding/main/source/encoded_frame.h

@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+
+#include "module_common_types.h"
+#include "common_types.h"
+#include "video_coding_defines.h"
+#include "video_image.h"
+
+namespace webrtc
+{
+
+class VCMEncodedFrame : protected EncodedImage
+{
+public:
+    VCMEncodedFrame();
+    VCMEncodedFrame(const webrtc::EncodedImage& rhs);
+    VCMEncodedFrame(const VCMEncodedFrame& rhs);
+
+    ~VCMEncodedFrame();
+    /**
+    *   Delete VideoFrame and resets members to zero
+    */
+    void Free();
+    /**
+    *   Set render time in milliseconds
+    */
+    void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
+
+    /**
+    *   Set the encoded frame size
+    */
+    void SetEncodedSize(WebRtc_UWord32 width, WebRtc_UWord32 height)
+                       { _encodedWidth  = width; _encodedHeight = height; }
+    /**
+    *   Get the encoded image
+    */
+    const webrtc::EncodedImage& EncodedImage() const
+                       { return static_cast<const webrtc::EncodedImage&>(*this); }
+    /**
+    *   Get pointer to frame buffer
+    */
+    const WebRtc_UWord8* Buffer() const {return _buffer;}
+    /**
+    *   Get frame length
+    */
+    WebRtc_UWord32 Length() const {return _length;}
+    /**
+    *   Get frame timestamp (90kHz)
+    */
+    WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
+    /**
+    *   Get render time in milliseconds
+    */
+    WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
+    /**
+    *   Get frame type
+    */
+    webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
+    /**
+    *   True if this frame is complete, false otherwise
+    */
+    bool Complete() const { return _completeFrame; }
+    /**
+    *   True if there's a frame missing before this frame
+    */
+    bool MissingFrame() const { return _missingFrame; }
+    /**
+    *   Payload type of the encoded payload
+    */
+    WebRtc_UWord8 PayloadType() const { return _payloadType; }
+    /**
+    *   Get codec specific info
+    */
+    const void* CodecSpecificInfo() const {return _codecSpecificInfo;}
+
+    WebRtc_Word32 Store(VCMFrameStorageCallback& storeCallback) const;
+
+    static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
+    static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
+
+protected:
+    /**
+    * Verifies that current allocated buffer size is larger than or equal to the input size.
+    * If the current buffer size is smaller, a new allocation is made and the old buffer data
+    * is copied to the new buffer.
+    * Buffer size is updated to minimumSize.
+    */
+    WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
+
+    void Reset();
+
+    WebRtc_Word64                 _renderTimeMs;
+    WebRtc_UWord8                 _payloadType;
+    bool                          _missingFrame;
+    void*                         _codecSpecificInfo;
+    WebRtc_UWord32                _codecSpecificInfoLength;
+    webrtc::VideoCodecType        _codec;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_

modules/video_coding/main/source/event.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
+#define WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
+
+#include "event_wrapper.h"
+
+namespace webrtc
+{
+
+//#define EVENT_DEBUG
+
+class VCMEvent : public EventWrapper
+{
+public:
+    VCMEvent() : _event(*EventWrapper::Create()) {};
+
+    virtual ~VCMEvent() { delete &_event; };
+
+    /**
+    *   Release waiting threads
+    */
+    bool Set() { return _event.Set(); };
+
+    bool Reset() { return _event.Reset(); };
+
+    /**
+    *   Wait for this event
+    */
+    EventTypeWrapper Wait(unsigned long maxTime)
+    {
+#ifdef EVENT_DEBUG
+        return kEventTimeout;
+#else
+        return _event.Wait(maxTime);
+#endif
+    };
+
+    /**
+    *   Start a timer
+    */
+    bool StartTimer(bool periodic, unsigned long time)
+                   { return _event.StartTimer(periodic, time); };
+    /**
+    *   Stop the timer
+    */
+    bool StopTimer() { return _event.StopTimer(); };
+
+private:
+    EventWrapper&      _event;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_EVENT_H_

modules/video_coding/main/source/exp_filter.cc

@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "exp_filter.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+void
+VCMExpFilter::Reset(float alpha)
+{
+    _alpha = alpha;
+    _filtered = -1.0;
+}
+
+float
+VCMExpFilter::Apply(float exp, float sample)
+{
+    if (_filtered == -1.0)
+    {
+        // Initialize filtered bit rates
+        _filtered = sample;
+    }
+    else if (exp == 1.0)
+    {
+        _filtered = _alpha * _filtered + (1 - _alpha) * sample;
+    }
+    else
+    {
+        float alpha = pow(_alpha, exp);
+        _filtered = alpha * _filtered + (1 - alpha) * sample;
+    }
+    if (_max != -1 && _filtered > _max)
+    {
+        _filtered = _max;
+    }
+    return _filtered;
+}
+
+void
+VCMExpFilter::UpdateBase(float alpha)
+{
+    _alpha = alpha;
+}
+
+float
+VCMExpFilter::Value() const
+{
+    return _filtered;
+}
+
+}

modules/video_coding/main/source/exp_filter.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
+
+namespace webrtc
+{
+
+/**********************/
+/* ExpFilter class    */
+/**********************/
+
+class VCMExpFilter
+{
+public:
+    VCMExpFilter(float alpha, float max = -1.0) : _alpha(alpha), _filtered(-1.0), _max(max) {}
+
+    // Resets the filter to its initial state, and resets alpha to the given value
+    //
+    // Input:
+    //          - alpha     : the new value of the filter factor base.
+    void Reset(float alpha);
+
+    // Applies the filter with the given exponent on the provided sample
+    //
+    // Input:
+    //          - exp       : Exponent T in y(k) = alpha^T * y(k-1) + (1 - alpha^T) * x(k)
+    //          - sample    : x(k) in the above filter equation
+    float Apply(float exp, float sample);
+
+    // Return current filtered value: y(k)
+    //
+    // Return value         : The current filter output
+    float Value() const;
+
+    // Change the filter factor base
+    //
+    // Input:
+    //          - alpha     : The new filter factor base.
+    void UpdateBase(float alpha);
+
+private:
+    float          _alpha;     // Filter factor base
+    float          _filtered;  // Current filter output
+    const float    _max;
+}; // end of ExpFilter class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_

modules/video_coding/main/source/frame_buffer.cc

@@ -0,0 +1,370 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "../../../../engine_configurations.h"
+#include "frame_buffer.h"
+#include "packet.h"
+
+#include <cassert>
+#include <string.h>
+
+#if defined(_WIN32)
+    // VS 2005: Don't warn for default initialized arrays. See help for more info.
+    #pragma warning(disable:4351)
+#endif
+
+namespace webrtc {
+
+// Constructor
+VCMFrameBuffer::VCMFrameBuffer() :
+    _state(kStateFree),
+    _frameCounted(false),
+    _nackCount(0),
+    _latestPacketTimeMs(-1)
+{
+}
+
+// Destructor
+VCMFrameBuffer::~VCMFrameBuffer()
+{
+    Reset();
+}
+
+VCMFrameBuffer::VCMFrameBuffer(VCMFrameBuffer& rhs)
+:
+VCMEncodedFrame(rhs),
+_state(rhs._state),
+_frameCounted(rhs._frameCounted),
+_sessionInfo(),
+_nackCount(rhs._nackCount),
+_latestPacketTimeMs(rhs._latestPacketTimeMs)
+{
+    _sessionInfo = rhs._sessionInfo;
+}
+
+webrtc::FrameType
+VCMFrameBuffer::FrameType() const
+{
+    return _sessionInfo.FrameType();
+}
+
+void
+VCMFrameBuffer::SetPreviousFrameLoss()
+{
+    _sessionInfo.SetPreviousFrameLoss();
+}
+
+WebRtc_Word32
+VCMFrameBuffer::GetLowSeqNum()
+{
+    return _sessionInfo.GetLowSeqNum();
+}
+
+// Get highest sequence number for complete sessions
+WebRtc_Word32
+VCMFrameBuffer::GetHighSeqNumComplete()
+{
+    if (_sessionInfo.IsSessionComplete())
+    {
+        return _sessionInfo.GetHighSeqNum();
+    }
+    return -1;
+}
+
+WebRtc_Word32
+VCMFrameBuffer::GetHighSeqNum()
+{
+    return _sessionInfo.GetHighSeqNum();
+}
+
+bool
+VCMFrameBuffer::IsSessionComplete()
+{
+    return _sessionInfo.IsSessionComplete();
+}
+
+// Insert packet
+VCMFrameBufferEnum
+VCMFrameBuffer::InsertPacket(const VCMPacket& packet, WebRtc_Word64 timeInMs)
+{
+    if (_state == kStateDecoding)
+    {
+        // Do not insert packet
+        return kIncomplete;
+    }
+
+    // Sanity to check if the frame has been freed. (Too old for example)
+    if(_state == kStateFree)
+    {
+        return kStateError;
+    }
+
+    // is this packet part of this frame
+    if (TimeStamp() && (TimeStamp() != packet.timestamp))
+    {
+        return kTimeStampError;
+    }
+
+    // sanity checks
+    if (_size + packet.sizeBytes + (packet.insertStartCode?kH264StartCodeLengthBytes:0) >
+            kMaxJBFrameSizeBytes)
+    {
+        return kSizeError;
+    }
+    if (NULL == packet.dataPtr && packet.sizeBytes > 0)
+    {
+        return kSizeError;
+    }
+    if(!_sessionInfo.HaveStartSeqNumber())
+    {
+        _sessionInfo.SetStartSeqNumber(packet.seqNum);
+    }
+    if (packet.dataPtr != NULL)
+    {
+        _payloadType = packet.payloadType;
+    }
+
+    if (kStateEmpty == _state)
+    {
+        // This is the first packet inserted into this frame,
+        // store some info and set some initial values.
+        _timeStamp = packet.timestamp;
+        _codec = packet.codec;
+        SetState(kStateIncomplete);
+    }
+
+    WebRtc_UWord32 requiredSizeBytes = Length() + packet.sizeBytes + (packet.insertStartCode?kH264StartCodeLengthBytes:0);
+    if (requiredSizeBytes >= _size)
+    {
+        const WebRtc_UWord32 increments = requiredSizeBytes / kBufferIncStepSizeBytes +
+                                        (requiredSizeBytes % kBufferIncStepSizeBytes > 0);
+        const WebRtc_UWord32 newSize = _size + increments * kBufferIncStepSizeBytes;
+        if (newSize > kMaxJBFrameSizeBytes)
+        {
+            return kSizeError;
+        }
+        if (VerifyAndAllocate(newSize) == -1)
+        {
+            return kSizeError;
+        }
+    }
+    WebRtc_Word64 retVal = _sessionInfo.InsertPacket(packet, _buffer);
+    if(retVal == -1)
+    {
+        return kSizeError;
+    }
+    else if (retVal == -2)
+    {
+        return kDuplicatePacket;
+    }
+    // update length
+    _length = Length() + static_cast<WebRtc_UWord32>(retVal);
+
+    _latestPacketTimeMs = timeInMs;
+
+    if(_sessionInfo.IsSessionComplete())
+    {
+        return kCompleteSession;
+    }
+    else
+    {
+        // this layer is not complete
+        if (_state == kStateComplete)
+        {
+            // we already have a complete layer
+            // wait for all independent layers belonging to the same frame
+            _state = kStateIncomplete;
+        }
+    }
+    return kIncomplete;
+}
+
+WebRtc_Word64 VCMFrameBuffer::LatestPacketTimeMs()
+{
+    return _latestPacketTimeMs;
+}
+
+// Zero out all entries in list up to and including the (first) entry equal to _lowSeqNum
+WebRtc_Word32 VCMFrameBuffer::ZeroOutSeqNum(WebRtc_Word32* list, WebRtc_Word32 num)
+{
+    if(_sessionInfo.ZeroOutSeqNum(list, num) != 0)
+    {
+       return -1;
+    }
+    return 0;
+}
+
+void VCMFrameBuffer::IncrementNackCount()
+{
+    _nackCount++;
+}
+
+WebRtc_Word16 VCMFrameBuffer::GetNackCount() const
+{
+    return _nackCount;
+}
+
+bool VCMFrameBuffer::HaveLastPacket()
+{
+    return _sessionInfo.HaveLastPacket();
+}
+
+bool
+VCMFrameBuffer::ForceSetHaveLastPacket()
+{
+    _sessionInfo.ForceSetHaveLastPacket();
+    return _sessionInfo.IsSessionComplete();
+}
+
+void VCMFrameBuffer::Reset()
+{
+    _length = 0;
+    _timeStamp = 0;
+
+    _sessionInfo.Reset();
+    _frameCounted = false;
+    _payloadType = 0;
+    _nackCount = 0;
+    _latestPacketTimeMs = -1;
+    _state = kStateFree;
+    VCMEncodedFrame::Reset();
+}
+
+// Makes sure the session contain a decodable stream.
+void
+VCMFrameBuffer::MakeSessionDecodable()
+{
+    WebRtc_Word32 retVal = _sessionInfo.MakeSessionDecodable(_buffer);
+    // update length
+    _length -= retVal;
+}
+
+// Set state of frame
+void
+VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state)
+{
+    if(_state == state)
+    {
+        return;
+    }
+    switch (state)
+    {
+    case kStateFree:
+        // Reset everything
+        // We can go to this state from all other states.
+        // The one setting the state to free must ensure
+        // that the frame is removed from the timestamp
+        // ordered frame list in the jb.
+        Reset();
+        break;
+
+    case kStateIncomplete:
+        // we can go to this state from state kStateEmpty
+        assert(_state == kStateEmpty ||
+            _state == kStateDecoding);
+
+        // Do nothing, we received a packet
+        break;
+
+    case kStateComplete:
+        assert(_state == kStateEmpty ||
+               _state == kStateIncomplete);
+
+        break;
+
+    case kStateEmpty:
+        assert(_state == kStateFree);
+        // Do nothing
+        break;
+
+    case kStateDecoding:
+        // we can go to this state from state kStateComplete kStateIncomplete
+        assert(_state == kStateComplete || _state == kStateIncomplete);
+        // Transfer frame information to EncodedFrame and create any codec specific information
+        RestructureFrameInformation();
+        break;
+
+    default:
+        // Should never happen
+        assert(!"FrameBuffer::SetState Incorrect frame buffer state as input");
+        return;
+    }
+    _state = state;
+}
+
+void
+VCMFrameBuffer::RestructureFrameInformation()
+{
+    PrepareForDecode();
+    _frameType = ConvertFrameType(_sessionInfo.FrameType());
+    _completeFrame = _sessionInfo.IsSessionComplete();
+    _missingFrame = _sessionInfo.PreviousFrameLoss();
+}
+
+WebRtc_Word32
+VCMFrameBuffer::ExtractFromStorage(const EncodedVideoData& frameFromStorage)
+{
+    _frameType = ConvertFrameType(frameFromStorage.frameType);
+    _timeStamp = frameFromStorage.timeStamp;
+    _payloadType = frameFromStorage.payloadType;
+    _encodedWidth = frameFromStorage.encodedWidth;
+    _encodedHeight = frameFromStorage.encodedHeight;
+    _missingFrame = frameFromStorage.missingFrame;
+    _completeFrame = frameFromStorage.completeFrame;
+    _renderTimeMs = frameFromStorage.renderTimeMs;
+    _codec = frameFromStorage.codec;
+    if (VerifyAndAllocate(frameFromStorage.payloadSize) < 0)
+    {
+        return VCM_MEMORY;
+    }
+    memcpy(_buffer, frameFromStorage.payloadData, frameFromStorage.payloadSize);
+    _length = frameFromStorage.payloadSize;
+    return VCM_OK;
+}
+
+// Set counted status (as counted by JB or not)
+void VCMFrameBuffer::SetCountedFrame(bool frameCounted)
+{
+    _frameCounted = frameCounted;
+}
+
+bool VCMFrameBuffer::GetCountedFrame()
+{
+    return _frameCounted;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum
+VCMFrameBuffer::GetState() const
+{
+    return _state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum
+VCMFrameBuffer::GetState(WebRtc_UWord32& timeStamp) const
+{
+    timeStamp = TimeStamp();
+    return GetState();
+}
+
+bool
+VCMFrameBuffer::IsRetransmitted()
+{
+    return _sessionInfo.IsRetransmitted();
+}
+
+void
+VCMFrameBuffer::PrepareForDecode()
+{
+    _length = _sessionInfo.PrepareForDecode(_buffer, _codec);
+}
+
+}

modules/video_coding/main/source/frame_buffer.h

@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+#include "encoded_frame.h"
+#include "frame_list.h"
+#include "jitter_buffer_common.h"
+#include "session_info.h"
+
+namespace webrtc
+{
+
+class VCMFrameBuffer : public VCMEncodedFrame
+{
+public:
+    VCMFrameBuffer();
+    virtual ~VCMFrameBuffer();
+
+    VCMFrameBuffer(VCMFrameBuffer& rhs);
+
+    virtual void Reset();
+
+    VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, WebRtc_Word64 timeInMs);
+
+    // State
+    // Get current state of frame
+    VCMFrameBufferStateEnum GetState() const;
+    // Get current state and timestamp of frame
+    VCMFrameBufferStateEnum GetState(WebRtc_UWord32& timeStamp) const;
+    void SetState(VCMFrameBufferStateEnum state); // Set state of frame
+
+    bool IsRetransmitted();
+    bool IsSessionComplete();
+    bool HaveLastPacket();
+    bool ForceSetHaveLastPacket();
+    // Makes sure the session contain a decodable stream.
+    void MakeSessionDecodable();
+
+    // Sequence numbers
+    // Get lowest packet sequence number in frame
+    WebRtc_Word32 GetLowSeqNum();
+    // Get highest packet sequence number in frame
+    WebRtc_Word32 GetHighSeqNum();
+
+    // Get highest sequence number of complete session
+    WebRtc_Word32 GetHighSeqNumComplete();
+
+    // Set counted status (as counted by JB or not)
+    void SetCountedFrame(bool frameCounted);
+    bool GetCountedFrame();
+
+    // NACK
+    // Zero out all entries in list up to and including the entry equal to _lowSeqNum
+    WebRtc_Word32 ZeroOutSeqNum(WebRtc_Word32* list, WebRtc_Word32 num);
+    void IncrementNackCount();
+    WebRtc_Word16 GetNackCount() const;
+
+    WebRtc_Word64 LatestPacketTimeMs();
+
+    webrtc::FrameType FrameType() const;
+    void SetPreviousFrameLoss();
+
+    WebRtc_Word32 ExtractFromStorage(const EncodedVideoData& frameFromStorage);
+
+protected:
+    void RestructureFrameInformation();
+    void PrepareForDecode();
+
+private:
+    VCMFrameBufferStateEnum    _state;         // Current state of the frame
+    bool                       _frameCounted;  // If this frame has been counted by JB
+    VCMSessionInfo             _sessionInfo;
+    WebRtc_UWord16             _nackCount;
+    WebRtc_Word64              _latestPacketTimeMs;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_

modules/video_coding/main/source/frame_dropper.cc

@@ -0,0 +1,331 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "frame_dropper.h"
+#include "internal_defines.h"
+#include "trace.h"
+
+namespace webrtc
+{
+
+VCMFrameDropper::VCMFrameDropper(WebRtc_Word32 vcmId)
+:
+_vcmId(vcmId),
+_keyFrameSizeAvgKbits(0.9f),
+_keyFrameRatio(0.99f),
+_dropRatio(0.9f, 0.96f)
+{
+    Reset();
+}
+
+void
+VCMFrameDropper::Reset()
+{
+    _keyFrameRatio.Reset(0.99f);
+    _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
+    _keyFrameSizeAvgKbits.Reset(0.9f);
+    _keyFrameCount = 0;
+    _accumulator = 0.0f;
+    _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
+    _targetBitRate = 300.0f;
+    _userFrameRate = 30;
+    _keyFrameSpreadFrames = 0.5f * _userFrameRate;
+    _dropNext = false;
+    _dropRatio.Reset(0.9f);
+    _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
+    _dropCount = 0;
+    _windowSize = 0.5f;
+    _wasBelowMax = true;
+    _enabled = true;
+    _fastMode = false; // start with normal (non-aggressive) mode
+}
+
+void
+VCMFrameDropper::Enable(bool enable)
+{
+    _enabled = enable;
+}
+
+void
+VCMFrameDropper::Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame)
+{
+    if (!_enabled)
+    {
+        return;
+    }
+    float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
+    if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
+    {
+        _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
+        _keyFrameRatio.Apply(1.0, 1.0);
+        if (frameSizeKbits > _keyFrameSizeAvgKbits.Value())
+        {
+            // Remove the average key frame size since we
+            // compensate for key frames when adding delta
+            // frames.
+            frameSizeKbits -= _keyFrameSizeAvgKbits.Value();
+        }
+        else
+        {
+            // Shouldn't be negative, so zero is the lower bound.
+            frameSizeKbits = 0;
+        }
+        if (_keyFrameRatio.Value() > 1e-5 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
+        {
+            // We are sending key frames more often than our upper bound for
+            // how much we allow the key frame compensation to be spread
+            // out in time. Therefor we must use the key frame ratio rather
+            // than keyFrameSpreadFrames.
+            _keyFrameCount = static_cast<WebRtc_Word32>(1 / _keyFrameRatio.Value() + 0.5);
+        }
+        else
+        {
+            // Compensate for the key frame the following frames
+            _keyFrameCount = static_cast<WebRtc_Word32>(_keyFrameSpreadFrames + 0.5);
+        }
+    }
+    else
+    {
+        // Decrease the keyFrameRatio
+        _keyFrameRatio.Apply(1.0, 0.0);
+    }
+    // Change the level of the accumulator (bucket)
+    _accumulator += frameSizeKbits;
+}
+
+void
+VCMFrameDropper::Leak(WebRtc_UWord32 inputFrameRate)
+{
+    if (!_enabled)
+    {
+        return;
+    }
+    if (inputFrameRate < 1)
+    {
+        return;
+    }
+    if (_targetBitRate < 0.0f)
+    {
+        return;
+    }
+    _keyFrameSpreadFrames = 0.5f * inputFrameRate;
+    // T is the expected bits per frame (target). If all frames were the same size,
+    // we would get T bits per frame. Notice that T is also weighted to be able to
+    // force a lower frame rate if wanted.
+    float T = _targetBitRate / inputFrameRate;
+    if (_keyFrameCount > 0)
+    {
+        // Perform the key frame compensation
+        if (_keyFrameRatio.Value() > 0 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
+        {
+            T -= _keyFrameSizeAvgKbits.Value() * _keyFrameRatio.Value();
+        }
+        else
+        {
+            T -= _keyFrameSizeAvgKbits.Value() / _keyFrameSpreadFrames;
+        }
+        _keyFrameCount--;
+    }
+    _accumulator -= T;
+    UpdateRatio();
+
+}
+
+void
+VCMFrameDropper::UpdateNack(WebRtc_UWord32 nackBytes)
+{
+    if (!_enabled)
+    {
+        return;
+    }
+    _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
+}
+
+void
+VCMFrameDropper::FillBucket(float inKbits, float outKbits)
+{
+    _accumulator += (inKbits - outKbits);
+}
+
+void
+VCMFrameDropper::UpdateRatio()
+{
+    if (_accumulator > 1.3f * _accumulatorMax)
+    {
+        // Too far above accumulator max, react faster
+        _dropRatio.UpdateBase(0.8f);
+    }
+    else
+    {
+        // Go back to normal reaction
+        _dropRatio.UpdateBase(0.9f);
+    }
+    if (_accumulator > _accumulatorMax)
+    {
+        // We are above accumulator max, and should ideally
+        // drop a frame. Increase the dropRatio and drop
+        // the frame later.
+        if (_wasBelowMax)
+        {
+            _dropNext = true;
+        }
+        if (_fastMode)
+        {
+            // always drop in aggressive mode
+            _dropNext = true;
+        }
+
+        _dropRatio.Apply(1.0f, 1.0f);
+        _dropRatio.UpdateBase(0.9f);
+    }
+    else
+    {
+        _dropRatio.Apply(1.0f, 0.0f);
+    }
+    if (_accumulator < 0.0f)
+    {
+        _accumulator = 0.0f;
+    }
+    _wasBelowMax = _accumulator < _accumulatorMax;
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId),  "FrameDropper: dropRatio = %f accumulator = %f, accumulatorMax = %f", _dropRatio.Value(), _accumulator, _accumulatorMax);
+}
+
+// This function signals when to drop frames to the caller. It makes use of the dropRatio
+// to smooth out the drops over time.
+bool
+VCMFrameDropper::DropFrame()
+{
+    if (!_enabled)
+    {
+        return false;
+    }
+    if (_dropNext)
+    {
+        _dropNext = false;
+        _dropCount = 0;
+    }
+
+    if (_dropRatio.Value() >= 0.5f) // Drops per keep
+    {
+        // limit is the number of frames we should drop between each kept frame
+        // to keep our drop ratio. limit is positive in this case.
+        float denom = 1.0f - _dropRatio.Value();
+        if (denom < 1e-5)
+        {
+            denom = (float)1e-5;
+        }
+        WebRtc_Word32 limit = static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
+        if (_dropCount < 0)
+        {
+            // Reset the _dropCount since it was negative and should be positive.
+            if (_dropRatio.Value() > 0.4f)
+            {
+                _dropCount = -_dropCount;
+            }
+            else
+            {
+                _dropCount = 0;
+            }
+        }
+        if (_dropCount < limit)
+        {
+            // As long we are below the limit we should drop frames.
+            _dropCount++;
+            return true;
+        }
+        else
+        {
+            // Only when we reset _dropCount a frame should be kept.
+            _dropCount = 0;
+            return false;
+        }
+    }
+    else if (_dropRatio.Value() > 0.0f && _dropRatio.Value() < 0.5f) // Keeps per drop
+    {
+        // limit is the number of frames we should keep between each drop
+        // in order to keep the drop ratio. limit is negative in this case,
+        // and the _dropCount is also negative.
+        float denom = _dropRatio.Value();
+        if (denom < 1e-5)
+        {
+            denom = (float)1e-5;
+        }
+        WebRtc_Word32 limit = -static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
+        if (_dropCount > 0)
+        {
+            // Reset the _dropCount since we have a positive
+            // _dropCount, and it should be negative.
+            if (_dropRatio.Value() < 0.6f)
+            {
+                _dropCount = -_dropCount;
+            }
+            else
+            {
+                _dropCount = 0;
+            }
+        }
+        if (_dropCount > limit)
+        {
+            if (_dropCount == 0)
+            {
+                // Drop frames when we reset _dropCount.
+                _dropCount--;
+                return true;
+            }
+            else
+            {
+                // Keep frames as long as we haven't reached limit.
+                _dropCount--;
+                return false;
+            }
+        }
+        else
+        {
+            _dropCount = 0;
+            return false;
+        }
+    }
+    _dropCount = 0;
+    return false;
+
+    // A simpler version, unfiltered and quicker
+    //bool dropNext = _dropNext;
+    //_dropNext = false;
+    //return dropNext;
+}
+
+void
+VCMFrameDropper::SetRates(float bitRate, float userFrameRate)
+{
+    // Bit rate of -1 means infinite bandwidth.
+    _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
+    if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
+    {
+        // Rescale the accumulator level if the accumulator max decreases
+        _accumulator = bitRate / _targetBitRate * _accumulator;
+    }
+    _targetBitRate = bitRate;
+    if (userFrameRate > 0.0f)
+    {
+        _userFrameRate = userFrameRate;
+    }
+}
+
+float
+VCMFrameDropper::ActualFrameRate(WebRtc_UWord32 inputFrameRate) const
+{
+    if (!_enabled)
+    {
+        return static_cast<float>(inputFrameRate);
+    }
+    return inputFrameRate * (1.0f - _dropRatio.Value());
+}
+
+}

modules/video_coding/main/source/frame_dropper.h

@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
+
+#include "exp_filter.h"
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+/******************************/
+/* VCMFrameDropper class     */
+/****************************/
+// The Frame Dropper implements a variant of the leaky bucket algorithm
+// for keeping track of when to drop frames to avoid bit rate
+// over use when the encoder can't keep its bit rate.
+class VCMFrameDropper
+{
+public:
+    VCMFrameDropper(WebRtc_Word32 vcmId = 0);
+    // Resets the FrameDropper to its initial state.
+    // This means that the frameRateWeight is set to its
+    // default value as well.
+    void Reset();
+
+    void Enable(bool enable);
+    // Answers the question if it's time to drop a frame
+    // if we want to reach a given frame rate. Must be
+    // called for every frame.
+    //
+    // Return value     : True if we should drop the current frame
+    bool DropFrame();
+    // Updates the FrameDropper with the size of the latest encoded
+    // frame. The FrameDropper calculates a new drop ratio (can be
+    // seen as the probability to drop a frame) and updates its
+    // internal statistics.
+    //
+    // Input:
+    //          - frameSizeBytes    : The size of the latest frame
+    //                                returned from the encoder.
+    //          - deltaFrame        : True if the encoder returned
+    //                                a key frame.
+    void Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame);
+
+    void Leak(WebRtc_UWord32 inputFrameRate);
+
+    void UpdateNack(WebRtc_UWord32 nackBytes);
+
+    // Sets the target bit rate and the frame rate produced by
+    // the camera.
+    //
+    // Input:
+    //          - bitRate       : The target bit rate
+    void SetRates(float bitRate, float userFrameRate);
+
+    // Return value     : The current average frame rate produced
+    //                    if the DropFrame() function is used as
+    //                    instruction of when to drop frames.
+    float ActualFrameRate(WebRtc_UWord32 inputFrameRate) const;
+
+private:
+    void FillBucket(float inKbits, float outKbits);
+    void UpdateRatio();
+
+    WebRtc_Word32     _vcmId;
+    VCMExpFilter       _keyFrameSizeAvgKbits;
+    VCMExpFilter       _keyFrameRatio;
+    float           _keyFrameSpreadFrames;
+    WebRtc_Word32     _keyFrameCount;
+    float           _accumulator;
+    float           _accumulatorMax;
+    float           _targetBitRate;
+    bool            _dropNext;
+    VCMExpFilter       _dropRatio;
+    WebRtc_Word32     _dropCount;
+    float           _windowSize;
+    float           _userFrameRate;
+    bool            _wasBelowMax;
+    bool            _enabled;
+    bool            _fastMode;
+}; // end of VCMFrameDropper class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_

modules/video_coding/main/source/frame_list.cc

@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "frame_list.h"
+#include "frame_buffer.h"
+#include "jitter_buffer.h"
+#include <cstdlib>
+
+namespace webrtc {
+
+VCMFrameListTimestampOrderAsc::~VCMFrameListTimestampOrderAsc()
+{
+    Flush();
+}
+
+void
+VCMFrameListTimestampOrderAsc::Flush()
+{
+    while(Erase(First()) != -1) { }
+}
+
+// Inserts frame in timestamp order, with the oldest timestamp first. Takes wrap arounds into account
+WebRtc_Word32
+VCMFrameListTimestampOrderAsc::Insert(VCMFrameBuffer* frame)
+{
+    VCMFrameListItem* item = static_cast<VCMFrameListItem*>(First());
+    VCMFrameListItem* newItem = new VCMFrameListItem(frame);
+    bool inserted = false;
+    if (newItem == NULL)
+    {
+        return -1;
+    }
+    while (item != NULL)
+    {
+        const WebRtc_UWord32 itemTimestamp = item->GetItem()->TimeStamp();
+        if (VCMJitterBuffer::LatestTimestamp(itemTimestamp, frame->TimeStamp()) == itemTimestamp)
+        {
+            if (InsertBefore(item, newItem) < 0)
+            {
+                delete newItem;
+                return -1;
+            }
+            inserted = true;
+            break;
+        }
+        item = Next(item);
+    }
+    if (!inserted && ListWrapper::Insert(ListWrapper::Last(), newItem) < 0)
+    {
+        delete newItem;
+        return -1;
+    }
+    return 0;
+}
+
+VCMFrameBuffer*
+VCMFrameListTimestampOrderAsc::FirstFrame() const
+{
+    VCMFrameListItem* item = First();
+    if (item != NULL)
+    {
+        return item->GetItem();
+    }
+    return NULL;
+}
+
+VCMFrameListItem*
+VCMFrameListTimestampOrderAsc::FindFrameListItem(FindFrameCriteria criteria,
+                                         const void* compareWith,
+                                         VCMFrameListItem* startItem) const
+{
+    if (startItem == NULL)
+    {
+        startItem = First();
+    }
+    if (criteria == NULL)
+    {
+        return NULL;
+    }
+    while (startItem != NULL)
+    {
+        if (criteria(startItem->GetItem(), compareWith))
+        {
+            return startItem;
+        }
+        startItem = Next(startItem);
+    }
+    // No frame found
+    return NULL;
+}
+
+VCMFrameBuffer*
+VCMFrameListTimestampOrderAsc::FindFrame(FindFrameCriteria criteria,
+                                         const void* compareWith,
+                                         VCMFrameListItem* startItem) const
+{
+    const VCMFrameListItem* frameListItem = FindFrameListItem(criteria, compareWith, startItem);
+    if (frameListItem == NULL)
+    {
+        return NULL;
+    }
+    return frameListItem->GetItem();
+}
+
+}
+

modules/video_coding/main/source/frame_list.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_LIST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_LIST_H_
+
+#include "list_wrapper.h"
+#include "typedefs.h"
+#include <stdlib.h>
+
+namespace webrtc
+{
+
+class VCMFrameBuffer;
+
+typedef bool (*FindFrameCriteria)(VCMFrameBuffer*, const void*);
+
+class VCMFrameListItem : public ListItem
+{
+    friend class VCMFrameListTimestampOrderAsc;
+public:
+    VCMFrameListItem(const VCMFrameBuffer* ptr) : ListItem(ptr) {}
+    ~VCMFrameListItem() {};
+
+    VCMFrameBuffer* GetItem() const
+            { return static_cast<VCMFrameBuffer*>(ListItem::GetItem()); }
+};
+
+class VCMFrameListTimestampOrderAsc : public ListWrapper
+{
+public:
+    VCMFrameListTimestampOrderAsc() : ListWrapper() {};
+    ~VCMFrameListTimestampOrderAsc();
+
+    void Flush();
+
+    // Inserts frame in timestamp order, with the oldest timestamp first.
+    // Takes wrap arounds into account.
+    WebRtc_Word32 Insert(VCMFrameBuffer* frame);
+    VCMFrameBuffer* FirstFrame() const;
+    VCMFrameListItem* Next(VCMFrameListItem* item) const
+            { return static_cast<VCMFrameListItem*>(ListWrapper::Next(item)); }
+    VCMFrameListItem* Previous(VCMFrameListItem* item) const
+            { return static_cast<VCMFrameListItem*>(ListWrapper::Previous(item)); }
+    VCMFrameListItem* First() const
+            { return static_cast<VCMFrameListItem*>(ListWrapper::First()); }
+    VCMFrameListItem* Last() const
+            { return static_cast<VCMFrameListItem*>(ListWrapper::Last()); }
+    VCMFrameListItem* FindFrameListItem(FindFrameCriteria criteria,
+                              const void* compareWith = NULL,
+                              VCMFrameListItem* startItem = NULL) const;
+    VCMFrameBuffer* FindFrame(FindFrameCriteria criteria,
+                                             const void* compareWith = NULL,
+                                             VCMFrameListItem* startItem = NULL) const;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_LIST_H_

modules/video_coding/main/source/generic_decoder.cc

@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_coding.h"
+#include "trace.h"
+#include "generic_decoder.h"
+#include "internal_defines.h"
+#include "tick_time.h"
+
+namespace webrtc {
+
+VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing)
+:
+_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_receiveCallback(NULL),
+_timing(timing),
+_timestampMap(kDecoderFrameMemoryLength)
+{
+}
+
+VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
+{
+    delete &_critSect;
+}
+
+void VCMDecodedFrameCallback::SetUserReceiveCallback(VCMReceiveCallback* receiveCallback)
+{
+    CriticalSectionScoped cs(_critSect);
+    _receiveCallback = receiveCallback;
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Decoded(RawImage& decodedImage)
+{
+    CriticalSectionScoped cs(_critSect);
+    VCMFrameInformation* frameInfo = static_cast<VCMFrameInformation*>(_timestampMap.Pop(decodedImage._timeStamp));
+    if (frameInfo == NULL)
+    {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    WebRtc_Word32 ret = _timing.StopDecodeTimer(decodedImage._timeStamp, frameInfo->decodeStartTimeMs, VCMTickTime::MillisecondTimestamp());
+
+    if (_receiveCallback != NULL)
+    {
+        _frame.Swap(decodedImage._buffer, decodedImage._length, decodedImage._size);
+        _frame.SetWidth(decodedImage._width);
+        _frame.SetHeight(decodedImage._height);
+        _frame.SetTimeStamp(decodedImage._timeStamp);
+        _frame.SetRenderTime(frameInfo->renderTimeMs);
+        // Convert raw image to video frame
+        WebRtc_Word32 callbackReturn = _receiveCallback->FrameToRender(_frame);
+        if (callbackReturn < 0)
+        {
+            return callbackReturn;
+        }
+    }
+    if (ret < 0)
+    {
+        return ret;
+    }
+    return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebRtc_Word32
+VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
+{
+    CriticalSectionScoped cs(_critSect);
+    if (_receiveCallback != NULL)
+    {
+        return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
+    }
+    return -1;
+}
+
+WebRtc_Word32
+VCMDecodedFrameCallback::ReceivedDecodedFrame(const WebRtc_UWord64 pictureId)
+{
+    _lastReceivedPictureID = pictureId;
+    return 0;
+}
+
+WebRtc_UWord64 VCMDecodedFrameCallback::LastReceivedPictureID() const
+{
+    return _lastReceivedPictureID;
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo)
+{
+    CriticalSectionScoped cs(_critSect);
+    return _timestampMap.Add(timestamp, frameInfo);
+}
+
+WebRtc_Word32 VCMDecodedFrameCallback::Pop(WebRtc_UWord32 timestamp)
+{
+    CriticalSectionScoped cs(_critSect);
+    if (_timestampMap.Pop(timestamp) == NULL)
+    {
+        return VCM_GENERAL_ERROR;
+    }
+    return VCM_OK;
+}
+
+VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id, bool isExternal)
+:
+_id(id),
+_callback(NULL),
+_frameInfos(),
+_nextFrameInfoIdx(0),
+_decoder(decoder),
+_codecType(kVideoCodecUnknown),
+_isExternal(isExternal),
+_requireKeyFrame(false),
+_keyFrameDecoded(false)
+{
+}
+
+VCMGenericDecoder::~VCMGenericDecoder()
+{
+}
+
+WebRtc_Word32 VCMGenericDecoder::InitDecode(const VideoCodec* settings, WebRtc_Word32 numberOfCores, bool requireKeyFrame)
+{
+    _requireKeyFrame = requireKeyFrame;
+    _keyFrameDecoded = false;
+    _codecType = settings->codecType;
+
+    return _decoder.InitDecode(settings, numberOfCores);
+}
+
+WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame)
+{
+    if (_requireKeyFrame &&
+        !_keyFrameDecoded &&
+        frame.FrameType() != kVideoFrameKey &&
+        frame.FrameType() != kVideoFrameGolden)
+    {
+        // Require key frame is enabled, meaning that one key frame must be decoded
+        // before we can decode delta frames.
+        return VCM_CODEC_ERROR;
+    }
+    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = VCMTickTime::MillisecondTimestamp();
+    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
+    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
+
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_id),
+               "Decoding timestamp %u",
+               frame.TimeStamp());
+
+    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
+
+    WebRtc_Word32 ret = _decoder.Decode(frame.EncodedImage(),
+                                        frame.MissingFrame(),
+                                        frame.CodecSpecificInfo(),
+                                        frame.RenderTimeMs());
+
+    if (ret < WEBRTC_VIDEO_CODEC_OK)
+    {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id), "Decoder error: %d\n", ret);
+        _callback->Pop(frame.TimeStamp());
+        return ret;
+    }
+    // Update the key frame decoded variable so that we know whether or not we've decoded a key frame since reset.
+    _keyFrameDecoded = (frame.FrameType() == kVideoFrameKey || frame.FrameType() == kVideoFrameGolden);
+    return ret;
+}
+
+WebRtc_Word32
+VCMGenericDecoder::Release()
+{
+    _keyFrameDecoded = false;
+    return _decoder.Release();
+}
+
+WebRtc_Word32 VCMGenericDecoder::Reset()
+{
+    _keyFrameDecoded = false;
+    return _decoder.Reset();
+}
+
+WebRtc_Word32 VCMGenericDecoder::SetCodecConfigParameters(const WebRtc_UWord8* buffer, WebRtc_Word32 size)
+{
+    return _decoder.SetCodecConfigParameters(buffer, size);
+}
+
+WebRtc_Word32 VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
+{
+    _callback = callback;
+    return _decoder.RegisterDecodeCompleteCallback(callback);
+}
+
+bool VCMGenericDecoder::External() const
+{
+    return _isExternal;
+}
+
+}

modules/video_coding/main/source/generic_decoder.h

@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+
+#include "timing.h"
+#include "timestamp_map.h"
+#include "video_codec_interface.h"
+#include "encoded_frame.h"
+#include "module_common_types.h"
+
+namespace webrtc
+{
+
+class VCMReceiveCallback;
+
+enum { kDecoderFrameMemoryLength = 10 };
+
+struct VCMFrameInformation
+{
+    WebRtc_Word64     renderTimeMs;
+    WebRtc_Word64     decodeStartTimeMs;
+    void*             userData;
+};
+
+class VCMDecodedFrameCallback : public DecodedImageCallback
+{
+public:
+    VCMDecodedFrameCallback(VCMTiming& timing);
+    virtual ~VCMDecodedFrameCallback();
+    void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
+
+    virtual WebRtc_Word32 Decoded(RawImage& decodedImage);
+    virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
+    virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
+
+    WebRtc_UWord64 LastReceivedPictureID() const;
+
+    WebRtc_Word32 Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo);
+    WebRtc_Word32 Pop(WebRtc_UWord32 timestamp);
+
+private:
+    CriticalSectionWrapper&        _critSect;
+    VideoFrame                  _frame;
+    VCMReceiveCallback*         _receiveCallback;
+    VCMTiming&                  _timing;
+    VCMTimestampMap             _timestampMap;
+    WebRtc_UWord64				_lastReceivedPictureID;
+};
+
+
+class VCMGenericDecoder
+{
+    friend class VCMCodecDataBase;
+public:
+    VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id = 0, bool isExternal = false);
+    ~VCMGenericDecoder();
+
+    /**
+    *	Initialize the decoder with the information from the VideoCodec
+    */
+    WebRtc_Word32 InitDecode(const VideoCodec* settings,
+                             WebRtc_Word32 numberOfCores,
+                             bool requireKeyFrame);
+
+    /**
+    *	Decode to a raw I420 frame,
+    *
+    *	inputVideoBuffer	reference to encoded video frame
+    */
+    WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame);
+
+    /**
+    *	Free the decoder memory
+    */
+    WebRtc_Word32 Release();
+
+    /**
+    *	Reset the decoder state, prepare for a new call
+    */
+    WebRtc_Word32 Reset();
+
+    /**
+    *	Codec configuration data sent out-of-band, i.e. in SIP call setup
+    *
+    *	buffer pointer to the configuration data
+    *	size the size of the configuration data in bytes
+    */
+    WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/,
+                                           WebRtc_Word32 /*size*/);
+
+    WebRtc_Word32 RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
+
+    bool External() const;
+
+protected:
+
+    WebRtc_Word32               _id;
+    VCMDecodedFrameCallback*    _callback;
+    VCMFrameInformation         _frameInfos[kDecoderFrameMemoryLength];
+    WebRtc_UWord32              _nextFrameInfoIdx;
+    VideoDecoder&               _decoder;
+    VideoCodecType              _codecType;
+    bool                        _isExternal;
+    bool                        _requireKeyFrame;
+    bool                        _keyFrameDecoded;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_

modules/video_coding/main/source/generic_encoder.cc

@@ -0,0 +1,229 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "encoded_frame.h"
+#include "generic_encoder.h"
+#include "media_optimization.h"
+#include "../../../../engine_configurations.h"
+
+namespace webrtc {
+
+//#define DEBUG_ENCODER_BIT_STREAM
+
+VCMGenericEncoder::VCMGenericEncoder(VideoEncoder& encoder, bool internalSource /*= false*/)
+:
+_encoder(encoder),
+_codecType(kVideoCodecUnknown),
+_VCMencodedFrameCallback(NULL),
+_bitRate(0),
+_frameRate(0),
+_internalSource(false)
+{
+}
+
+
+VCMGenericEncoder::~VCMGenericEncoder()
+{
+}
+
+WebRtc_Word32
+VCMGenericEncoder::Reset()
+{
+    _bitRate = 0;
+    _frameRate = 0;
+    _VCMencodedFrameCallback = NULL;
+    return _encoder.Reset();
+}
+
+WebRtc_Word32 VCMGenericEncoder::Release()
+{
+    _bitRate = 0;
+    _frameRate = 0;
+    _VCMencodedFrameCallback = NULL;
+    return _encoder.Release();
+}
+
+WebRtc_Word32
+VCMGenericEncoder::InitEncode(const VideoCodec* settings, WebRtc_Word32 numberOfCores, WebRtc_UWord32 maxPayloadSize)
+{
+    _bitRate = settings->startBitrate;
+    _frameRate = settings->maxFramerate;
+    _codecType = settings->codecType;
+    if (_VCMencodedFrameCallback != NULL)
+    {
+        _VCMencodedFrameCallback->SetCodecType(_codecType);
+    }
+    return _encoder.InitEncode(settings, numberOfCores, maxPayloadSize);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::Encode(const VideoFrame& inputFrame, const void* codecSpecificInfo, FrameType frameType)
+{
+    RawImage rawImage(inputFrame.Buffer(), inputFrame.Length(), inputFrame.Size());
+    rawImage._width     = inputFrame.Width();
+    rawImage._height    = inputFrame.Height();
+    rawImage._timeStamp = inputFrame.TimeStamp();
+
+    WebRtc_Word32 ret = _encoder.Encode(rawImage, codecSpecificInfo, VCMEncodedFrame::ConvertFrameType(frameType));
+
+    return ret;
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetPacketLoss(WebRtc_Word32 packetLoss)
+{
+    return _encoder.SetPacketLoss(packetLoss);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate)
+{
+    WebRtc_Word32 ret = _encoder.SetRates(newBitRate, frameRate);
+    if (ret < 0)
+    {
+        return ret;
+    }
+    _bitRate = newBitRate;
+    _frameRate = frameRate;
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMGenericEncoder::CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size)
+{
+    WebRtc_Word32 ret = _encoder.CodecConfigParameters(buffer, size);
+    if (ret < 0)
+    {
+        return ret;
+    }
+    return ret;
+}
+
+WebRtc_UWord32 VCMGenericEncoder::BitRate() const
+{
+    return _bitRate;
+}
+
+WebRtc_UWord32 VCMGenericEncoder::FrameRate() const
+{
+    return _frameRate;
+}
+
+WebRtc_Word32
+VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
+{
+    return _encoder.SetPeriodicKeyFrames(enable);
+}
+
+WebRtc_Word32
+VCMGenericEncoder::RequestFrame(FrameType frameType)
+{
+    RawImage image;
+    return _encoder.Encode(image, NULL, VCMEncodedFrame::ConvertFrameType(frameType));
+}
+
+WebRtc_Word32
+VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback)
+{
+   _VCMencodedFrameCallback = VCMencodedFrameCallback;
+
+   _VCMencodedFrameCallback->SetCodecType(_codecType);
+   _VCMencodedFrameCallback->SetInternalSource(_internalSource);
+   return _encoder.RegisterEncodeCompleteCallback(_VCMencodedFrameCallback);
+}
+
+bool
+VCMGenericEncoder::InternalSource() const
+{
+    return _internalSource;
+}
+
+ /***************************
+  * Callback Implementation
+  ***************************/
+VCMEncodedFrameCallback::VCMEncodedFrameCallback():
+_sendCallback(),
+_encodedBytes(0),
+_payloadType(0),
+_bitStreamAfterEncoder(NULL)
+{
+#ifdef DEBUG_ENCODER_BIT_STREAM
+    _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
+#endif
+}
+
+VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
+{
+#ifdef DEBUG_ENCODER_BIT_STREAM
+    fclose(_bitStreamAfterEncoder);
+#endif
+}
+
+WebRtc_Word32
+VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
+{
+    _sendCallback = transport;
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMEncodedFrameCallback::Encoded(EncodedImage &encodedImage, const void* codecSpecificInfo,
+                                 const RTPFragmentationHeader* fragmentationHeader)
+{
+    FrameType frameType = VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
+
+    WebRtc_UWord32 encodedBytes = 0;
+    if (_sendCallback != NULL)
+    {
+            encodedBytes = encodedImage._length;
+
+        if (_bitStreamAfterEncoder != NULL)
+        {
+            fwrite(encodedImage._buffer, 1, encodedImage._length, _bitStreamAfterEncoder);
+        }
+
+        WebRtc_Word32 callbackReturn = _sendCallback->SendData(frameType,
+                                                               _payloadType,
+                                                               encodedImage._timeStamp,
+                                                               encodedImage._buffer,
+                                                               encodedBytes,
+                                                               *fragmentationHeader);
+       if (callbackReturn < 0)
+       {
+           return callbackReturn;
+       }
+    }
+    else
+    {
+        return VCM_UNINITIALIZED;
+    }
+    _encodedBytes = encodedBytes;
+    _mediaOpt->UpdateWithEncodedData(_encodedBytes, frameType);
+    if (_internalSource)
+    {
+        return _mediaOpt->DropFrame(); // Signal to encoder to drop next frame
+    }
+
+    return VCM_OK;
+}
+
+WebRtc_UWord32
+VCMEncodedFrameCallback::EncodedBytes()
+{
+    return _encodedBytes;
+}
+
+void
+VCMEncodedFrameCallback::SetMediaOpt(VCMMediaOptimization *mediaOpt)
+{
+    _mediaOpt = mediaOpt;
+}
+
+}

modules/video_coding/main/source/generic_encoder.h

@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+
+#include "video_codec_interface.h"
+
+#include <stdio.h>
+
+namespace webrtc
+{
+
+class VCMMediaOptimization;
+
+/*************************************/
+/* VCMEncodeFrameCallback class     */
+/***********************************/
+class VCMEncodedFrameCallback : public EncodedImageCallback
+{
+public:
+    VCMEncodedFrameCallback();
+    virtual ~VCMEncodedFrameCallback();
+
+    /*
+    * Callback implementation - codec encode complete
+    */
+    WebRtc_Word32 Encoded(EncodedImage& encodedImage, const void* codecSpecificInfo = NULL,
+                          const RTPFragmentationHeader* fragmentationHeader = NULL);
+    /*
+    * Get number of encoded bytes
+    */
+    WebRtc_UWord32 EncodedBytes();
+    /*
+    * Callback implementation - generic encoder encode complete
+    */
+    WebRtc_Word32 SetTransportCallback(VCMPacketizationCallback* transport);
+    /**
+    * Set media Optimization
+    */
+    void SetMediaOpt (VCMMediaOptimization* mediaOpt);
+
+    void SetPayloadType(WebRtc_UWord8 payloadType) { _payloadType = payloadType; };
+    void SetCodecType(VideoCodecType codecType) {_codecType = codecType;};
+    void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
+
+private:
+    VCMPacketizationCallback* _sendCallback;
+    VCMMediaOptimization*           _mediaOpt;
+    WebRtc_UWord32                  _encodedBytes;
+    WebRtc_UWord8                   _payloadType;
+    VideoCodecType              _codecType;
+    bool                            _internalSource;
+    FILE*                           _bitStreamAfterEncoder;
+
+};// end of VCMEncodeFrameCallback class
+
+
+/******************************/
+/* VCMGenericEncoder class    */
+/******************************/
+class VCMGenericEncoder
+{
+    friend class VCMCodecDataBase;
+public:
+    VCMGenericEncoder(VideoEncoder& encoder, bool internalSource = false);
+    ~VCMGenericEncoder();
+    /**
+    *	Reset the encoder state, prepare for a new call
+    */
+    WebRtc_Word32 Reset();
+    /**
+    *	Free encoder memory
+    */
+    WebRtc_Word32 Release();
+    /**
+    *	Initialize the encoder with the information from the VideoCodec
+    */
+    WebRtc_Word32 InitEncode(const VideoCodec* settings,
+                             WebRtc_Word32 numberOfCores,
+                             WebRtc_UWord32 maxPayloadSize);
+    /**
+    *	Encode raw image
+    *	inputFrame        : Frame containing raw image
+    *	codecSpecificInfo : Specific codec data
+    *	cameraFrameRate	  :	request or information from the remote side
+    *	frameType         : The requested frame type to encode
+    */
+    WebRtc_Word32 Encode(const VideoFrame& inputFrame,
+                         const void* codecSpecificInfo,
+                         FrameType frameType);
+    /**
+    *	Set new target bit rate and frame rate
+    * Return Value: new bit rate if OK, otherwise <0s
+    */
+    WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate);
+    /**
+    * Set a new packet loss rate
+    */
+    WebRtc_Word32 SetPacketLoss(WebRtc_Word32 packetLoss);
+    WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size);
+    /**
+    * Register a transport callback which will be called to deliver the encoded buffers
+    */
+    WebRtc_Word32 RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback);
+    /**
+    * Get encoder bit rate
+    */
+    WebRtc_UWord32 BitRate() const;
+     /**
+    * Get encoder frame rate
+    */
+    WebRtc_UWord32 FrameRate() const;
+
+    WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
+
+    WebRtc_Word32 RequestFrame(FrameType frameType);
+
+    bool InternalSource() const;
+
+private:
+    VideoEncoder&               _encoder;
+    VideoCodecType              _codecType;
+    VCMEncodedFrameCallback*    _VCMencodedFrameCallback;
+    WebRtc_UWord32              _bitRate;
+    WebRtc_UWord32              _frameRate;
+    bool                        _internalSource;
+}; // end of VCMGenericEncoder class
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_

modules/video_coding/main/source/inter_frame_delay.cc

@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "inter_frame_delay.h"
+#include "tick_time.h"
+
+namespace webrtc {
+
+VCMInterFrameDelay::VCMInterFrameDelay()
+{
+    Reset();
+}
+
+// Resets the delay estimate
+void
+VCMInterFrameDelay::Reset()
+{
+    _zeroWallClock = VCMTickTime::MillisecondTimestamp();
+    _wrapArounds = 0;
+    _prevWallClock = 0;
+    _prevTimestamp = 0;
+    _dTS = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+bool
+VCMInterFrameDelay::CalculateDelay(WebRtc_UWord32 timestamp,
+                                WebRtc_Word64 *delay,
+                                WebRtc_Word64 currentWallClock /* = -1 */)
+{
+    if (currentWallClock <= -1)
+    {
+        currentWallClock = VCMTickTime::MillisecondTimestamp();
+    }
+
+    if (_prevWallClock == 0)
+    {
+        // First set of data, initialization, wait for next frame
+        _prevWallClock = currentWallClock;
+        _prevTimestamp = timestamp;
+        *delay = 0;
+        return true;
+    }
+
+    WebRtc_Word32 prevWrapArounds = _wrapArounds;
+    CheckForWrapArounds(timestamp);
+
+    // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
+    WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+    // Account for reordering in jitter variance estimate in the future?
+    // Note that this also captures incomplete frames which are grabbed
+    // for decoding after a later frame has been complete, i.e. real
+    // packet losses.
+    if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
+    {
+        *delay = 0;
+        return false;
+    }
+
+    // Compute the compensated timestamp difference and convert it to ms and
+    // round it to closest integer.
+    _dTS = static_cast<WebRtc_Word64>((timestamp + wrapAroundsSincePrev *
+                (static_cast<WebRtc_Word64>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
+
+    // frameDelay is the difference of dT and dTS -- i.e. the difference of
+    // the wall clock time difference and the timestamp difference between
+    // two following frames.
+    *delay = static_cast<WebRtc_Word64>(currentWallClock - _prevWallClock - _dTS);
+
+    _prevTimestamp = timestamp;
+    _prevWallClock = currentWallClock;
+
+    return true;
+}
+
+// Returns the current difference between incoming timestamps
+WebRtc_UWord32 VCMInterFrameDelay::CurrentTimeStampDiffMs() const
+{
+    if (_dTS < 0)
+    {
+        return 0;
+    }
+    return static_cast<WebRtc_UWord32>(_dTS);
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp and
+// keeps track of the number of wrap arounds since reset.
+void
+VCMInterFrameDelay::CheckForWrapArounds(WebRtc_UWord32 timestamp)
+{
+    if (timestamp < _prevTimestamp)
+    {
+        // This difference will probably be less than -2^31 if we have had a wrap around
+        // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
+        // it should be positive.
+        if (static_cast<WebRtc_Word32>(timestamp - _prevTimestamp) > 0)
+        {
+            // Forward wrap around
+            _wrapArounds++;
+        }
+    }
+    // This difference will probably be less than -2^31 if we have had a backward wrap around.
+    // Since it is cast to a Word32, it should be positive.
+    else if (static_cast<WebRtc_Word32>(_prevTimestamp - timestamp) > 0)
+    {
+        // Backward wrap around
+        _wrapArounds--;
+    }
+}
+
+}

modules/video_coding/main/source/inter_frame_delay.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class VCMInterFrameDelay
+{
+public:
+    VCMInterFrameDelay();
+
+    // Resets the estimate. Zeros are given as parameters.
+    void Reset();
+
+    // Calculates the delay of a frame with the given timestamp.
+    // This method is called when the frame is complete.
+    //
+    // Input:
+    //          - timestamp         : RTP timestamp of a received frame
+    //          - *delay            : Pointer to memory where the result should be stored
+    //          - currentWallClock  : The current time in milliseconds.
+    //                                Should be -1 for normal operation, only used for testing.
+    // Return value                 : true if OK, false when reordered timestamps
+    bool CalculateDelay(WebRtc_UWord32 timestamp,
+                        WebRtc_Word64 *delay,
+                        WebRtc_Word64 currentWallClock = -1);
+
+    // Returns the current difference between incoming timestamps
+    //
+    // Return value                 : Wrap-around compensated difference between incoming
+    //                                timestamps.
+    WebRtc_UWord32 CurrentTimeStampDiffMs() const;
+
+private:
+    // Controls if the RTP timestamp counter has had a wrap around
+    // between the current and the previously received frame.
+    //
+    // Input:
+    //          - timestmap         : RTP timestamp of the current frame.
+    void CheckForWrapArounds(WebRtc_UWord32 timestamp);
+
+    WebRtc_Word64         _zeroWallClock; // Local timestamp of the first video packet received
+    WebRtc_Word32         _wrapArounds;   // Number of wrapArounds detected
+    // The previous timestamp passed to the delay estimate
+    WebRtc_UWord32        _prevTimestamp;
+    // The previous wall clock timestamp used by the delay estimate
+    WebRtc_Word64         _prevWallClock;
+    // Wrap-around compensated difference between incoming timestamps
+    WebRtc_Word64         _dTS;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_

modules/video_coding/main/source/internal_defines.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
+
+inline WebRtc_UWord32 MaskWord64ToUWord32(WebRtc_Word64 w64)
+{
+    return static_cast<WebRtc_UWord32>(MASK_32_BITS(w64));
+}
+
+#define VCM_MAX(a, b) ((a) > (b)) ? (a) : (b)
+#define VCM_MIN(a, b) ((a) < (b)) ? (a) : (b)
+
+#define VCM_DEFAULT_CODEC_WIDTH 352
+#define VCM_DEFAULT_CODEC_HEIGHT 288
+#define VCM_DEFAULT_FRAME_RATE 30
+#define VCM_MIN_BITRATE 30
+
+// Helper macros for creating the static codec list
+#define VCM_NO_CODEC_IDX -1
+#ifdef VIDEOCODEC_VP8
+  #define VCM_VP8_IDX VCM_NO_CODEC_IDX + 1
+#else
+  #define VCM_VP8_IDX VCM_NO_CODEC_IDX
+#endif
+#ifdef VIDEOCODEC_I420
+  #define VCM_I420_IDX VCM_VP8_IDX + 1
+#else
+  #define VCM_I420_IDX VCM_VP8_IDX
+#endif
+#define VCM_NUM_VIDEO_CODECS_AVAILABLE VCM_I420_IDX + 1
+
+#define VCM_NO_RECEIVER_ID 0
+
+inline WebRtc_Word32 VCMId(const WebRtc_Word32 vcmId, const WebRtc_Word32 receiverId = 0)
+{
+    return static_cast<WebRtc_Word32>((vcmId << 16) + receiverId);
+}
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_

modules/video_coding/main/source/jitter_buffer.h

@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+
+#include "typedefs.h"
+#include "critical_section_wrapper.h"
+#include "module_common_types.h"
+#include "video_coding_defines.h"
+#include "inter_frame_delay.h"
+#include "event.h"
+#include "frame_list.h"
+#include "jitter_buffer_common.h"
+#include "jitter_estimator.h"
+
+namespace webrtc
+{
+
+// forward declarations
+class VCMFrameBuffer;
+class VCMPacket;
+class VCMEncodedFrame;
+
+class VCMJitterSample
+{
+public:
+    VCMJitterSample() : timestamp(0), frameSize(0), latestPacketTime(-1) {}
+    WebRtc_UWord32 timestamp;
+    WebRtc_UWord32 frameSize;
+    WebRtc_Word64 latestPacketTime;
+};
+
+class VCMJitterBuffer
+{
+public:
+    VCMJitterBuffer(WebRtc_Word32 vcmId = -1,
+                    WebRtc_Word32 receiverId = -1,
+                    bool master = true);
+    virtual ~VCMJitterBuffer();
+
+    VCMJitterBuffer& operator=(const VCMJitterBuffer& rhs);
+
+    // We need a start and stop to break out of the wait event
+    // used in GetCompleteFrameForDecoding
+    void Start();
+    void Stop();
+    bool Running() const;
+
+    // Empty the Jitter buffer of all its data
+    void Flush();
+
+    // Statistics, Get received key and delta frames
+    WebRtc_Word32 GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
+                                     WebRtc_UWord32& receivedKeyFrames) const;
+
+    // Statistics, Calculate frame and bit rates
+    WebRtc_Word32 GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate);
+
+    // Wait for the first packet in the next frame to arrive, blocks for <= maxWaitTimeMS ms
+    WebRtc_Word64 GetNextTimeStamp(WebRtc_UWord32 maxWaitTimeMS,
+                                   FrameType& incomingFrameType,
+                                   WebRtc_Word64& renderTimeMs);
+
+    // Will the packet sequence be complete if the next frame is grabbed
+    // for decoding right now? That is, have we lost a frame between the
+    // last decoded frame and the next, or is the next frame missing one
+    // or more packets?
+    bool CompleteSequenceWithNextFrame();
+
+    // Wait maxWaitTimeMS for a complete frame to arrive. After timeout NULL is returned.
+    VCMEncodedFrame* GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS);
+
+    // Get a frame for decoding (even an incomplete) without delay.
+    VCMEncodedFrame* GetFrameForDecoding();
+
+    VCMEncodedFrame* GetFrameForDecodingNACK();
+
+    // Release frame (when done with decoding)
+    void ReleaseFrame(VCMEncodedFrame* frame);
+
+    // Get frame to use for this timestamp
+    WebRtc_Word32 GetFrame(const VCMPacket& packet, VCMEncodedFrame*&);
+    VCMEncodedFrame* GetFrame(const VCMPacket& packet); // deprecated
+
+    // Returns the time in ms when the latest packet was insterted into the frame.
+    // Retransmitted is set to true if any of the packets belonging to the frame
+    // has been retransmitted.
+    WebRtc_Word64 LastPacketTime(VCMEncodedFrame* frame, bool& retransmitted) const;
+
+    // Insert a packet into a frame
+    VCMFrameBufferEnum InsertPacket(VCMEncodedFrame* frame, const VCMPacket& packet);
+
+    // Sync
+    WebRtc_UWord32 GetEstimatedJitterMS();
+    void UpdateRtt(WebRtc_UWord32 rttMs);
+
+    // NACK
+    void SetNackStatus(bool enable); // Enable/disable nack
+    bool GetNackStatus();            // Get nack status (enabled/disabled)
+    // Get list of missing sequence numbers (size in number of elements)
+    WebRtc_UWord16* GetNackList(WebRtc_UWord16& nackSize, bool& listExtended);
+
+    WebRtc_Word64 LastDecodedTimestamp() const;
+    static WebRtc_UWord32 LatestTimestamp(const WebRtc_UWord32 existingTimestamp,
+                                          const WebRtc_UWord32 newTimestamp);
+
+protected:
+
+    // Misc help functions
+    // Recycle (release) frame, used if we didn't receive whole frame
+    void RecycleFrame(VCMFrameBuffer* frame);
+    void ReleaseFrameInternal(VCMFrameBuffer* frame);
+    // Flush and reset the jitter buffer. Call under critical section.
+    void FlushInternal();
+    VCMFrameListItem* FindOldestSequenceNum() const;
+
+    // Help functions for insert packet
+    // Get empty frame, creates new (i.e. increases JB size) if necessary
+    VCMFrameBuffer* GetEmptyFrame();
+    // Recycle oldest frames up to a key frame, used if JB is completely full
+    bool RecycleFramesUntilKeyFrame();
+    // Update frame state (set as complete or reconstructable if conditions are met)
+    void UpdateFrameState(VCMFrameBuffer* frameListItem);
+
+    // Help functions for getting a frame
+    // Find oldest complete frame, used for getting next frame to decode
+    VCMFrameListItem* FindOldestCompleteContinuousFrame();
+
+    // Check if a frame is missing the markerbit but is complete
+    bool CheckForCompleteFrame(VCMFrameListItem* oldestFrameItem);
+
+
+    void CleanUpOldFrames();
+    void CleanUpSizeZeroFrames();
+
+    void VerifyAndSetPreviousFrameLost(VCMFrameBuffer& frame);
+    bool IsPacketRetransmitted(const VCMPacket& packet) const;
+    void UpdateJitterAndDelayEstimates(VCMJitterSample& sample, bool incompleteFrame);
+    void UpdateJitterAndDelayEstimates(VCMFrameBuffer& frame, bool incompleteFrame);
+    void UpdateJitterAndDelayEstimates(WebRtc_Word64 latestPacketTimeMs,
+                                       WebRtc_UWord32 timestamp,
+                                       WebRtc_UWord32 frameSize,
+                                       bool incompleteFrame);
+    void UpdateOldJitterSample(const VCMPacket& packet);
+    WebRtc_UWord32 GetEstimatedJitterMsInternal();
+
+    // NACK help
+    WebRtc_UWord16* CreateNackList(WebRtc_UWord16& nackSize, bool& listExtended);
+    WebRtc_Word32 GetLowHighSequenceNumbers(WebRtc_Word32& lowSeqNum,
+                                            WebRtc_Word32& highSeqNum) const;
+
+    void UpdateLastDecodedWithFiller(const VCMPacket& packet);
+
+private:
+
+    static bool FrameEqualTimestamp(VCMFrameBuffer* frame, const void* timestamp);
+    static bool CompleteKeyFrameCriteria(VCMFrameBuffer* frame, const void* notUsed);
+
+    WebRtc_Word32                 _vcmId;
+    WebRtc_Word32                 _receiverId;
+    bool                          _running;     // If we are running (have started) or not
+    CriticalSectionWrapper&          _critSect;
+    bool                          _master;
+    // Event to signal when we have a frame ready for decoder
+    VCMEvent                      _frameEvent;
+    // Event to signal when we have received a packet
+    VCMEvent                      _packetEvent;
+    WebRtc_Word32                 _maxNumberOfFrames; // Number of allocated frames
+    // Array of pointers to the frames in JB
+    VCMFrameBuffer*               _frameBuffers[kMaxNumberOfFrames];
+    VCMFrameListTimestampOrderAsc _frameBuffersTSOrder;
+
+    // timing
+    // Sequence number of last frame that was given to decoder
+    WebRtc_Word32           _lastDecodedSeqNum;
+    // Timestamp of last frame that was given to decoder
+    WebRtc_Word64           _lastDecodedTimeStamp;
+
+    // Statistics
+    // Frame counter for each type (key, delta, golden, key-delta)
+    WebRtc_UWord8           _receiveStatistics[4];
+    // Latest calculated frame rates of incoming stream
+    WebRtc_UWord8           _incomingFrameRate;
+    WebRtc_UWord32          _incomingFrameCount;   // Frame counter, reset in GetUpdate
+    // Real time for last _frameCount reset
+    WebRtc_Word64           _timeLastIncomingFrameCount;
+    WebRtc_UWord32          _incomingBitCount;     // Received bits counter, reset in GetUpdate
+    WebRtc_UWord32          _incomingBitRate;
+    WebRtc_UWord32          _dropCount;            // Frame drop counter
+    // Number of frames in a row that have been too old
+    WebRtc_UWord32          _numConsecutiveOldFrames;
+    // Number of packets in a row that have been too old
+    WebRtc_UWord32          _numConsecutiveOldPackets;
+    // Filters for estimating jitter
+    VCMJitterEstimator      _jitterEstimate;
+    // Calculates network delays used for jitter calculations
+    VCMInterFrameDelay      _delayEstimate;
+    VCMJitterSample         _waitingForCompletion;
+
+    // NACK
+    bool                    _usingNACK; // If we are using nack
+    // Holds the internal nack list (the missing seqence numbers)
+    WebRtc_Word32           _NACKSeqNumInternal[kNackHistoryLength];
+    WebRtc_UWord16          _NACKSeqNum[kNackHistoryLength];
+    WebRtc_UWord32          _NACKSeqNumLength;
+
+    bool                    _missingMarkerBits;
+    bool                    _firstPacket;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_

modules/video_coding/main/source/jitter_buffer_common.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+
+namespace webrtc
+{
+
+enum { kMaxNumberOfFrames     = 100 };
+enum { kStartNumberOfFrames   = 6 };    // in packets, 6 packets are approximately 198 ms,
+                                        // we need at least one more for process
+enum { kMaxVideoDelayMs       = 2000 }; // in ms
+
+enum VCMJitterBufferEnum
+{
+    kMaxConsecutiveOldFrames        = 60,
+    kMaxConsecutiveOldPackets       = 300,
+    kMaxPacketsInJitterBuffer       = 800,
+    kBufferIncStepSizeBytes         = 30000,       // >20 packets
+    kMaxJBFrameSizeBytes            = 4000000      // sanity don't go above 4Mbyte
+};
+
+enum VCMFrameBufferEnum
+{
+    kStateError           = -4,
+    kTimeStampError       = -2,
+    kSizeError            = -1,
+    kNoError              = 0,
+    kIncomplete           = 1,    // Frame incomplete
+    kFirstPacket          = 2,
+    kCompleteSession      = 3,    // at least one layer in the frame complete
+    kDuplicatePacket      = 5     // We're receiving a duplicate packet.
+};
+
+enum VCMFrameBufferStateEnum
+{
+    kStateFree,               // Unused frame in the JB
+    kStateEmpty,              // frame popped by the RTP receiver
+    kStateIncomplete,         // frame that have one or more packet(s) stored
+    kStateComplete,           // frame that have all packets
+    kStateDecoding            // frame popped by the decoding thread
+};
+
+enum { kH264StartCodeLengthBytes = 4};
+
+// Used to indicate if a received packet contain a complete NALU (or equivalent)
+enum VCMNaluCompleteness
+{
+    kNaluUnset=0,       //Packet has not been filled.
+    kNaluComplete=1,    //Packet can be decoded as is.
+    kNaluStart,         // Packet contain beginning of NALU
+    kNaluIncomplete,    //Packet is not beginning or end of NALU
+    kNaluEnd            // Packet is the end of a NALU
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_

modules/video_coding/main/source/jitter_estimator.cc

@@ -0,0 +1,439 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "jitter_estimator.h"
+#include "rtt_filter.h"
+#include "tick_time.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+namespace webrtc {
+
+VCMJitterEstimator::VCMJitterEstimator(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId) :
+_vcmId(vcmId),
+_receiverId(receiverId),
+_phi(0.97),
+_psi(0.9999),
+_alphaCountMax(400),
+_beta(0.9994),
+_thetaLow(0.000001),
+_nackLimit(3), // This should be 1 if the old
+               // retransmition estimate is used.
+_nackWindowMS(15000),
+_numStdDevDelayOutlier(15),
+_numStdDevFrameSizeOutlier(3),
+_noiseStdDevs(2.33), // ~Less than 1% chance
+                     // (look up in normal distribution table)...
+_noiseStdDevOffset(30.0), // ...of getting 30 ms freezes
+_rttFilter(vcmId, receiverId)
+{
+    Reset();
+}
+
+VCMJitterEstimator&
+VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs)
+{
+    if (this != &rhs)
+    {
+        memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
+        memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
+
+        _vcmId = rhs._vcmId;
+        _receiverId = rhs._receiverId;
+        _avgFrameSize = rhs._avgFrameSize;
+        _varFrameSize = rhs._varFrameSize;
+        _maxFrameSize = rhs._maxFrameSize;
+        _fsSum = rhs._fsSum;
+        _fsCount = rhs._fsCount;
+        _lastUpdateT = rhs._lastUpdateT;
+        _prevEstimate = rhs._prevEstimate;
+        _prevFrameSize = rhs._prevFrameSize;
+        _avgNoise = rhs._avgNoise;
+        _alphaCount = rhs._alphaCount;
+        _filterJitterEstimate = rhs._filterJitterEstimate;
+        _startupCount = rhs._startupCount;
+        _latestNackTimestamp = rhs._latestNackTimestamp;
+        _nackCount = rhs._nackCount;
+        _rttFilter = rhs._rttFilter;
+    }
+    return *this;
+}
+
+// Resets the JitterEstimate
+void
+VCMJitterEstimator::Reset()
+{
+    _theta[0] = 1/(512e3/8);
+    _theta[1] = 0;
+    _varNoise = 4.0;
+
+    _thetaCov[0][0] = 1e-4;
+    _thetaCov[1][1] = 1e2;
+    _thetaCov[0][1] = _thetaCov[1][0] = 0;
+    _Qcov[0][0] = 2.5e-10;
+    _Qcov[1][1] = 1e-10;
+    _Qcov[0][1] = _Qcov[1][0] = 0;
+    _avgFrameSize = 500;
+    _maxFrameSize = 500;
+    _varFrameSize = 100;
+    _lastUpdateT = -1;
+    _prevEstimate = -1.0;
+    _prevFrameSize = 0;
+    _avgNoise = 0.0;
+    _alphaCount = 1;
+    _filterJitterEstimate = 0.0;
+    _latestNackTimestamp = 0;
+    _nackCount = 0;
+    _fsSum = 0;
+    _fsCount = 0;
+    _startupCount = 0;
+    _rttFilter.Reset();
+}
+
+void
+VCMJitterEstimator::ResetNackCount()
+{
+    _nackCount = 0;
+}
+
+// Updates the estimates with the new measurements
+void
+VCMJitterEstimator::UpdateEstimate(WebRtc_Word64 frameDelayMS, WebRtc_UWord32 frameSizeBytes,
+                                            bool incompleteFrame /* = false */)
+{
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
+               VCMId(_vcmId, _receiverId),
+               "Jitter estimate updated with: frameSize=%d frameDelayMS=%d",
+               frameSizeBytes, frameDelayMS);
+    if (frameSizeBytes == 0)
+    {
+        return;
+    }
+    int deltaFS = frameSizeBytes - _prevFrameSize;
+    if (_fsCount < kFsAccuStartupSamples)
+    {
+        _fsSum += frameSizeBytes;
+        _fsCount++;
+    }
+    else if (_fsCount == kFsAccuStartupSamples)
+    {
+        // Give the frame size filter
+        _avgFrameSize = static_cast<double>(_fsSum) /
+                        static_cast<double>(_fsCount);
+        _fsCount++;
+    }
+    if (!incompleteFrame || frameSizeBytes > _avgFrameSize)
+    {
+        double avgFrameSize = _phi * _avgFrameSize +
+                              (1 - _phi) * frameSizeBytes;
+        if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize))
+        {
+            // Only update the average frame size if this sample wasn't a
+            // key frame
+            _avgFrameSize = avgFrameSize;
+        }
+        // Update the variance anyway since we want to capture cases where we only get
+        // key frames.
+        _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) *
+                                (frameSizeBytes - avgFrameSize) *
+                                (frameSizeBytes - avgFrameSize), 1.0);
+    }
+
+    // Update max frameSize estimate
+    _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
+
+    if (_prevFrameSize == 0)
+    {
+        _prevFrameSize = frameSizeBytes;
+        return;
+    }
+    _prevFrameSize = frameSizeBytes;
+
+    // Only update the Kalman filter if the sample is not considered
+    // an extreme outlier. Even if it is an extreme outlier from a
+    // delay point of view, if the frame size also is large the
+    // deviation is probably due to an incorrect line slope.
+    double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
+
+    if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
+        frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
+    {
+        // Update the variance of the deviation from the
+        // line given by the Kalman filter
+        EstimateRandomJitter(deviation, incompleteFrame);
+        // Prevent updating with frames which have been congested by a large
+        // frame, and therefore arrives almost at the same time as that frame.
+        // This can occur when we receive a large frame (key frame) which
+        // has been delayed. The next frame is of normal size (delta frame),
+        // and thus deltaFS will be << 0. This removes all frame samples
+        // which arrives after a key frame.
+        if ((!incompleteFrame || deviation >= 0.0) &&
+            static_cast<double>(deltaFS) > - 0.25 * _maxFrameSize)
+        {
+            // Update the Kalman filter with the new data
+            KalmanEstimateChannel(frameDelayMS, deltaFS);
+        }
+    }
+    else
+    {
+        int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
+        EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
+    }
+    // Post process the total estimated jitter
+    if (_startupCount >= kStartupDelaySamples)
+    {
+        PostProcessEstimate();
+    }
+    else
+    {
+        _startupCount++;
+    }
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "Framesize statistics: max=%f average=%f", _maxFrameSize, _avgFrameSize);
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "The estimated slope is: theta=(%f, %f)", _theta[0], _theta[1]);
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "Random jitter: mean=%f variance=%f", _avgNoise, _varNoise);
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "Current jitter estimate: %f", _filterJitterEstimate);
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "Current max RTT: %u", _rttFilter.RttMs());
+}
+
+// Updates the nack/packet ratio
+void
+VCMJitterEstimator::UpdateNackEstimate(bool retransmitted, WebRtc_Word64 /*wallClockMS = -1*/)
+{
+    // Simplified since it seems to be hard to be sure if a
+    // packet actually has been retransmitted or not, resulting
+    // in a delay which varies up and down with one RTT.
+    // The solution is to wait until _nackLimit retransmitts
+    // has been received, then always add an RTT to the estimate.
+    if (retransmitted && _nackCount < _nackLimit)
+    {
+        _nackCount++;
+    }
+    //if (wallClockMS == -1)
+    //{
+    //    wallClockMS = VCMTickTime::MillisecondTimestamp();
+    //}
+    //if (retransmitted)
+    //{
+    //    if (_nackCount < _nackLimit)
+    //    {
+    //        _nackCount++;
+    //    }
+    //    _latestNackTimestamp = wallClockMS;
+    //}
+    //else if (_nackCount > 0 && wallClockMS - _latestNackTimestamp > _nackWindowMS)
+    //{
+    //    _nackCount = 0;
+    //}
+}
+
+// Updates Kalman estimate of the channel
+// The caller is expected to sanity check the inputs.
+void
+VCMJitterEstimator::KalmanEstimateChannel(WebRtc_Word64 frameDelayMS,
+                                          WebRtc_Word32 deltaFSBytes)
+{
+    double Mh[2];
+    double hMh_sigma;
+    double kalmanGain[2];
+    double measureRes;
+    double t00, t01;
+
+    // Kalman filtering
+
+    // Prediction
+    // M = M + Q
+    _thetaCov[0][0] += _Qcov[0][0];
+    _thetaCov[0][1] += _Qcov[0][1];
+    _thetaCov[1][0] += _Qcov[1][0];
+    _thetaCov[1][1] += _Qcov[1][1];
+
+    // Kalman gain
+    // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
+    // h = [dFS 1]
+    // Mh = M*h'
+    // hMh_sigma = h*M*h' + R
+    Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
+    Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
+    // sigma weights measurements with a small deltaFS as noisy and
+    // measurements with large deltaFS as good
+    if (_maxFrameSize < 1.0)
+    {
+        return;
+    }
+    double sigma = (300.0 * exp(-abs(static_cast<double>(deltaFSBytes)) /
+                   (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
+    if (sigma < 1.0)
+    {
+        sigma = 1.0;
+    }
+    hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
+    if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0))
+    {
+        assert(false);
+        return;
+    }
+    kalmanGain[0] = Mh[0] / hMh_sigma;
+    kalmanGain[1] = Mh[1] / hMh_sigma;
+
+    // Correction
+    // theta = theta + K*(dT - h*theta)
+    measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
+    _theta[0] += kalmanGain[0] * measureRes;
+    _theta[1] += kalmanGain[1] * measureRes;
+
+    if (_theta[0] < _thetaLow)
+    {
+        _theta[0] = _thetaLow;
+    }
+
+    // M = (I - K*h)*M
+    t00 = _thetaCov[0][0];
+    t01 = _thetaCov[0][1];
+    _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
+                      kalmanGain[0] * _thetaCov[1][0];
+    _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
+                      kalmanGain[0] * _thetaCov[1][1];
+    _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
+                      kalmanGain[1] * deltaFSBytes * t00;
+    _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
+                      kalmanGain[1] * deltaFSBytes * t01;
+
+    // Covariance matrix, must be positive semi-definite
+    assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
+           _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 &&
+           _thetaCov[0][0] >= 0);
+}
+
+// Calculate difference in delay between a sample and the
+// expected delay estimated by the Kalman filter
+double
+VCMJitterEstimator::DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
+                                               WebRtc_Word32 deltaFSBytes) const
+{
+    return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
+}
+
+// Estimates the random jitter by calculating the variance of the
+// sample distance from the line given by theta.
+void
+VCMJitterEstimator::EstimateRandomJitter(double d_dT, bool incompleteFrame)
+{
+    double alpha;
+    if (_alphaCount == 0)
+    {
+        assert(_alphaCount > 0);
+        return;
+    }
+    alpha = static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
+    _alphaCount++;
+    if (_alphaCount > _alphaCountMax)
+    {
+        _alphaCount = _alphaCountMax;
+    }
+    double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
+    double varNoise = alpha * _varNoise +
+                      (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
+    if (!incompleteFrame || varNoise > _varNoise)
+    {
+        _avgNoise = avgNoise;
+        _varNoise = varNoise;
+    }
+    if (_varNoise < 1.0)
+    {
+        // The variance should never be zero, since we might get
+        // stuck and consider all samples as outliers.
+        _varNoise = 1.0;
+    }
+}
+
+double
+VCMJitterEstimator::NoiseThreshold() const
+{
+    double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
+    if (noiseThreshold < 1.0)
+    {
+        noiseThreshold = 1.0;
+    }
+    return noiseThreshold;
+}
+
+// Calculates the current jitter estimate from the filtered estimates
+double
+VCMJitterEstimator::CalculateEstimate()
+{
+    double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
+
+    // A very low estimate (or negative) is neglected
+    if (ret < 1.0) {
+        if (_prevEstimate <= 0.01)
+        {
+            ret = 1.0;
+        }
+        else
+        {
+            ret = _prevEstimate;
+        }
+    }
+    if (ret > 10000.0) // Sanity
+    {
+        ret = 10000.0;
+    }
+    _prevEstimate = ret;
+    return ret;
+}
+
+void
+VCMJitterEstimator::PostProcessEstimate()
+{
+    _filterJitterEstimate = CalculateEstimate();
+}
+
+void
+VCMJitterEstimator::UpdateRtt(WebRtc_UWord32 rttMs)
+{
+    _rttFilter.Update(rttMs);
+}
+
+void
+VCMJitterEstimator::UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes)
+{
+    if (_maxFrameSize < frameSizeBytes)
+    {
+        _maxFrameSize = frameSizeBytes;
+    }
+}
+
+// Returns the current filtered estimate if available,
+// otherwise tries to calculate an estimate.
+double
+VCMJitterEstimator::GetJitterEstimate()
+{
+    double jitterMS = CalculateEstimate();
+    if (_filterJitterEstimate > jitterMS)
+    {
+        jitterMS = _filterJitterEstimate;
+    }
+    if (_nackCount >= _nackLimit)
+    {
+        return jitterMS + _rttFilter.RttMs();
+    }
+    return jitterMS;
+}
+
+}

modules/video_coding/main/source/jitter_estimator.h

@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+
+#include "typedefs.h"
+#include "rtt_filter.h"
+
+namespace webrtc
+{
+
+class VCMJitterEstimator
+{
+public:
+    VCMJitterEstimator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
+
+    VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
+
+    // Resets the estimate to the initial state
+    void Reset();
+    void ResetNackCount();
+
+    // Updates the jitter estimate with the new data.
+    //
+    // Input:
+    //          - frameDelay      : Delay-delta calculated by UTILDelayEstimate in milliseconds
+    //          - frameSize       : Frame size of the current frame.
+    //          - incompleteFrame : Flags if the frame is used to update the estimate before it
+    //                              was complete. Default is false.
+    void UpdateEstimate(WebRtc_Word64 frameDelayMS,
+                        WebRtc_UWord32 frameSizeBytes,
+                        bool incompleteFrame = false);
+
+    // Returns the current jitter estimate in milliseconds and adds
+    // also adds an RTT dependent term in cases of retransmission.
+    //
+    // Return value                   : Jitter estimate in milliseconds
+    double GetJitterEstimate();
+
+    // Updates the nack counter/timer.
+    //
+    // Input:
+    //          - retransmitted       : True for a nacked frames, false otherwise
+    //          - wallClockMS         : Used for testing
+    void UpdateNackEstimate(bool retransmitted, WebRtc_Word64 wallClockMS = -1);
+
+    // Updates the RTT filter.
+    //
+    // Input:
+    //          - rttMs               : RTT in ms
+    void UpdateRtt(WebRtc_UWord32 rttMs);
+
+    void UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes);
+
+    // A constant describing the delay from the jitter buffer
+    // to the delay on the receiving side which is not accounted
+    // for by the jitter buffer nor the decoding delay estimate.
+    static const WebRtc_UWord32 OPERATING_SYSTEM_JITTER = 10;
+
+protected:
+    // These are protected for better testing possibilities
+    double              _theta[2]; // Estimated line parameters (slope, offset)
+    double              _varNoise; // Variance of the time-deviation from the line
+
+private:
+    // Updates the Kalman filter for the line describing
+    // the frame size dependent jitter.
+    //
+    // Input:
+    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
+    //          - deltaFSBytes    : Frame size delta, i.e.
+    //                            : frame size at time T minus frame size at time T-1
+    void KalmanEstimateChannel(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes);
+
+    // Updates the random jitter estimate, i.e. the variance
+    // of the time deviations from the line given by the Kalman filter.
+    //
+    // Input:
+    //          - d_dT              : The deviation from the kalman estimate
+    //          - incompleteFrame   : True if the frame used to update the estimate
+    //                                with was incomplete
+    void EstimateRandomJitter(double d_dT, bool incompleteFrame);
+
+    double NoiseThreshold() const;
+
+    // Calculates the current jitter estimate.
+    //
+    // Return value                 : The current jitter estimate in milliseconds
+    double CalculateEstimate();
+
+    // Post process the calculated estimate
+    void PostProcessEstimate();
+
+    // Calculates the difference in delay between a sample and the
+    // expected delay estimated by the Kalman filter.
+    //
+    // Input:
+    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
+    //          - deltaFS         : Frame size delta, i.e. frame size at time
+    //                              T minus frame size at time T-1
+    //
+    // Return value                 : The difference in milliseconds
+    double DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
+                                      WebRtc_Word32 deltaFSBytes) const;
+
+    // Constants, filter parameters
+    WebRtc_Word32         _vcmId;
+    WebRtc_Word32         _receiverId;
+    const double          _phi;
+    const double          _psi;
+    const WebRtc_UWord32  _alphaCountMax;
+    const double          _beta;
+    const double          _thetaLow;
+    const WebRtc_UWord32  _nackLimit;
+    const WebRtc_UWord32  _nackWindowMS;
+    const WebRtc_Word32   _numStdDevDelayOutlier;
+    const WebRtc_Word32   _numStdDevFrameSizeOutlier;
+    const double          _noiseStdDevs;
+    const double          _noiseStdDevOffset;
+
+    double                _thetaCov[2][2]; // Estimate covariance
+    double                _Qcov[2][2];     // Process noise covariance
+    double                _avgFrameSize;   // Average frame size
+    double                _varFrameSize;   // Frame size variance
+    double                _maxFrameSize;   // Largest frame size received (descending
+                                           // with a factor _psi)
+    WebRtc_UWord32        _fsSum;
+    WebRtc_UWord32        _fsCount;
+
+    WebRtc_Word64         _lastUpdateT;
+    double                _prevEstimate;         // The previously returned jitter estimate
+    WebRtc_UWord32        _prevFrameSize;        // Frame size of the previous frame
+    double                _avgNoise;             // Average of the random jitter
+    WebRtc_UWord32        _alphaCount;
+    double                _filterJitterEstimate; // The filtered sum of jitter estimates
+
+    WebRtc_UWord32        _startupCount;
+
+    WebRtc_Word64         _latestNackTimestamp;  // Timestamp in ms when the latest nack was seen
+    WebRtc_UWord32        _nackCount;            // Keeps track of the number of nacks received,
+                                                 // but never goes above _nackLimit
+    VCMRttFilter          _rttFilter;
+
+    enum { kStartupDelaySamples = 30 };
+    enum { kFsAccuStartupSamples = 5 };
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_

modules/video_coding/main/source/media_opt_util.cc

@@ -0,0 +1,850 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_coding_defines.h"
+#include "fec_tables_xor.h"
+#include "er_tables_xor.h"
+#include "nack_fec_tables.h"
+#include "qm_select_data.h"
+#include "media_opt_util.h"
+
+#include <math.h>
+#include <float.h>
+#include <limits.h>
+#include <stdio.h>
+
+namespace webrtc {
+
+bool
+VCMProtectionMethod::BetterThan(VCMProtectionMethod *pm)
+{
+    if (pm == NULL)
+    {
+        return true;
+    }
+    return pm->_score > _score;
+}
+
+bool
+VCMNackFecMethod::ProtectionFactor(const VCMProtectionParameters* /*parameters*/)
+{
+
+   //use FEC model with modification with RTT for now
+
+    return true;
+}
+
+bool
+VCMNackFecMethod::EffectivePacketLoss(const VCMProtectionParameters* /*parameters*/)
+{
+  //use FEC model with modification with RTT for now
+
+  return true;
+}
+
+
+bool
+VCMNackFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+
+     VCMFecMethod fecMethod;
+     VCMNackMethod nackMethod;
+
+    const WebRtc_UWord8 plossMax = 129;
+     WebRtc_UWord16 rttMax = nackMethod.MaxRttNack();
+
+    // We should reduce the NACK threshold for NackFec protection method,
+    // with FEC and ER, we should only use NACK for small RTT, to avoid delay
+     //But this parameter change should be shared with RTP and JB
+     //rttMax  = (WebRtc_UWord16) 0.5*rttMax;
+
+     //Compute the protection factor
+     fecMethod.ProtectionFactor(parameters);
+
+     //Compute the effective packet loss
+     fecMethod.EffectivePacketLoss(parameters);
+
+     WebRtc_UWord8 protFactorK = fecMethod._protectionFactorK;
+     WebRtc_UWord8 protFactorD = fecMethod._protectionFactorD;
+     WebRtc_UWord8 effPacketLoss = fecMethod._effectivePacketLoss;
+     float resPacketLoss = fecMethod._residualPacketLoss;
+
+    WebRtc_Word16 rttIndex= (WebRtc_UWord16) parameters->rtt;
+    float softnessRtt = 1.0;
+    if (parameters->rtt < rttMax)
+    {
+         softnessRtt = (float)VCMNackFecTable[rttIndex]/(float)4096.0;
+
+         //soften ER with NACK on
+        //table depends on roundtrip time relative to rttMax (NACK Threshold)
+        _effectivePacketLoss = (WebRtc_UWord8)(effPacketLoss*softnessRtt);
+
+        //soften FEC with NACK on
+        //table depends on roundtrip time relative to rttMax (NACK Threshold)
+        _protectionFactorK = (WebRtc_UWord8) (protFactorK * softnessRtt);
+        _protectionFactorD = (WebRtc_UWord8) (protFactorD * softnessRtt);
+    }
+
+
+    //make sure I frame protection is at least larger than P frame protection, and at least as high as received loss
+    WebRtc_UWord8 packetLoss = (WebRtc_UWord8)(255* parameters->lossPr);
+     _protectionFactorK = static_cast<WebRtc_UWord8>(VCM_MAX(packetLoss,VCM_MAX(_scaleProtKey*protFactorD,protFactorK)));
+
+    //check limit on amount of protection for I frame: 50% is max
+    if (_protectionFactorK  >= plossMax) _protectionFactorK  = plossMax - 1;
+
+    //Bit cost for NackFec
+
+     // NACK cost: based on residual packet loss (since we should only NACK packet not recovered by FEC)
+    _efficiency = 0.0f;
+    if (parameters->rtt < rttMax)
+        _efficiency = parameters->bitRate * resPacketLoss / (1.0f + resPacketLoss);
+
+    //add FEC cost: ignore I frames for now
+    float fecRate = static_cast<float>(_protectionFactorD) / 255.0f;
+    if (fecRate >= 0.0f)
+         _efficiency += parameters->bitRate * fecRate;
+
+    _score = _efficiency;
+
+    //Protection/fec rates obtained above is defined relative to total number of packets (total rate: source+fec)
+    //FEC in RTP module assumes protection factor is defined relative to source number of packets
+    //so we should convert the factor to reduce mismatch between mediaOpt suggested rate and the actual rate
+    WebRtc_UWord8 codeRate = protFactorK;
+    _protectionFactorK = fecMethod.ConvertFECRate(codeRate);
+    codeRate = protFactorD;
+    _protectionFactorD = fecMethod.ConvertFECRate(codeRate);
+
+    return true;
+}
+
+
+bool
+VCMNackMethod::EffectivePacketLoss(WebRtc_UWord8 effPacketLoss, WebRtc_UWord16 rttTime)
+{
+
+    WebRtc_UWord16 rttMax = MaxRttNack();
+
+    //For large RTT, we should rely on some Error Resilience, so we set packetLossEnc = 0
+    //for RTT less than the NACK threshold
+    if (rttTime < rttMax )
+        effPacketLoss =  0;   //may want a softer transition here
+
+     _effectivePacketLoss = effPacketLoss;
+
+    return true;
+
+}
+bool
+VCMNackMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+
+    //Compute the effective packet loss for ER
+    WebRtc_UWord8 effPacketLoss = (WebRtc_UWord8)(255* parameters->lossPr);
+    WebRtc_UWord16 rttTime = (WebRtc_UWord16) parameters->rtt;
+    EffectivePacketLoss(effPacketLoss, rttTime);
+    //
+
+    //Compute the NACK bit cost
+    _efficiency = parameters->bitRate * parameters->lossPr / (1.0f + parameters->lossPr);
+    _score = _efficiency;
+    if (parameters->rtt > _NACK_MAX_RTT)
+    {
+        _score = 0.0f;
+        return false;
+    }
+
+    return true;
+}
+
+
+WebRtc_UWord8
+VCMFecMethod::BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta, WebRtc_UWord8 packetFrameKey) const
+{
+
+    WebRtc_UWord8 boostRateKey = 2;
+    //default: ratio scales the FEC protection up for I frames
+    WebRtc_UWord8 ratio = 1;
+
+    if (packetFrameDelta > 0)
+        ratio = (WebRtc_Word8)( packetFrameKey / packetFrameDelta );
+
+    ratio = VCM_MAX(boostRateKey, ratio);
+
+    return ratio;
+
+}
+
+WebRtc_UWord8
+VCMFecMethod::ConvertFECRate(WebRtc_UWord8 codeRateRTP) const
+{
+    return static_cast<WebRtc_UWord8>(VCM_MIN(255,(0.5 + 255.0*codeRateRTP/(float)(255 - codeRateRTP))));
+}
+
+//AvgRecoveryFEC: average recovery from FEC, assuming random packet loss model
+//Computed offline for a range of FEC code parameters and loss rates
+float
+VCMFecMethod::AvgRecoveryFEC(const VCMProtectionParameters* parameters) const
+{
+
+    //Total (avg) bits available per frame: total rate over actual/sent frame rate
+    //units are kbits/frame
+    const WebRtc_UWord16 bitRatePerFrame = static_cast<WebRtc_UWord16>(parameters->bitRate/(parameters->frameRate));
+
+    //Total (avg) number of packets per frame (source and fec):
+    const WebRtc_UWord8 avgTotPackets  = 1 + (WebRtc_UWord8)((float)bitRatePerFrame*1000.0/(float)(8.0*_maxPayloadSize) + 0.5);
+
+    //parameters for tables
+    const WebRtc_UWord8 codeSize = 24;
+    const WebRtc_UWord8 plossMax = 129;
+    const WebRtc_UWord16 maxErTableSize = 38700;
+    //
+
+    //
+    //Get index for table
+    const float protectionFactor = (float)_protectionFactorD/(float)255;
+    WebRtc_UWord8 fecPacketsPerFrame =  (WebRtc_UWord8)(0.5 + protectionFactor*avgTotPackets);
+    WebRtc_UWord8 sourcePacketsPerFrame = avgTotPackets - fecPacketsPerFrame;
+
+    if (fecPacketsPerFrame == 0)
+    {
+        return 0.0; //no protection, so avg. recov from FEC == 0
+    }
+
+    //table defined up to codeSizexcodeSize code
+    if (sourcePacketsPerFrame > codeSize)
+    {
+        sourcePacketsPerFrame = codeSize;
+    }
+
+    //check: protection factor is maxed at 50%, so this should never happen
+    if (sourcePacketsPerFrame < 1)
+    {
+        assert("average number of source packets below 1\n");
+    }
+
+    //index for ER tables: up to codeSizexcodeSize mask
+    WebRtc_UWord16 codeIndexTable[codeSize*codeSize];
+    WebRtc_UWord16 k = -1;
+    for(WebRtc_UWord8 i=1;i<=codeSize;i++)
+    {
+        for(WebRtc_UWord8 j=1;j<=i;j++)
+        {
+            k += 1;
+            codeIndexTable[(j-1)*codeSize + i - 1] = k;
+        }
+    }
+
+    const WebRtc_UWord8 lossRate = (WebRtc_UWord8) (255.0*parameters->lossPr + 0.5f);
+
+    const WebRtc_UWord16 codeIndex = (fecPacketsPerFrame - 1)*codeSize + (sourcePacketsPerFrame - 1);
+    const WebRtc_UWord16 indexTable = codeIndexTable[codeIndex] * plossMax + lossRate;
+
+    const WebRtc_UWord16 codeIndex2 = (fecPacketsPerFrame)*codeSize + (sourcePacketsPerFrame);
+    WebRtc_UWord16 indexTable2 =  codeIndexTable[codeIndex2] * plossMax + lossRate;
+
+    //checks on table index
+    if (indexTable >= maxErTableSize)
+    {
+        assert("ER table index too large\n");
+    }
+
+    if (indexTable2 >= maxErTableSize)
+    {
+        indexTable2 = indexTable;
+    }
+    //
+
+    //Get the average effective packet loss recovery from FEC
+    //this is from tables, computed using random loss model
+    WebRtc_UWord8 avgFecRecov1 = 0;
+    WebRtc_UWord8 avgFecRecov2 = 0;
+    float avgFecRecov = 0;
+
+    if (fecPacketsPerFrame > 0)
+    {
+        avgFecRecov1 = VCMAvgFECRecoveryXOR[indexTable];
+        avgFecRecov2 = VCMAvgFECRecoveryXOR[indexTable2];
+    }
+
+    //interpolate over two FEC codes
+    const float weightRpl = (float)(0.5 + protectionFactor*avgTotPackets) - (float)fecPacketsPerFrame;
+    avgFecRecov = (float)weightRpl * (float)avgFecRecov2 + (float)(1.0 - weightRpl) * (float)avgFecRecov1;
+
+
+    return avgFecRecov;
+
+}
+
+bool
+VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
+{
+
+    //FEC PROTECTION SETTINGS: varies with packet loss and bitrate
+
+    const float bitRate = parameters->bitRate;
+    WebRtc_UWord8 packetLoss = (WebRtc_UWord8)(255* parameters->lossPr);
+
+
+    //Size of tables
+    const WebRtc_UWord16 maxFecTableSize =  6450;
+    //Parameters for range of rate and packet loss for tables
+    const WebRtc_UWord8 ratePar1 = 5;
+    const WebRtc_UWord8 ratePar2 = 49;
+    const WebRtc_UWord8 plossMax = 129;
+
+    //
+    //Just for testing: for the case where we randomly lose slices instead of RTP packets and use SingleMode packetization in RTP module
+    //const WebRtc_UWord16 slice_size = 3000/6; //corresponds to rate=1000k with 4 cores
+
+    //float slice_mtu = (float)_maxPayloadSize/(float)slice_size;
+    const float slice_mtu = 1.0;
+    //
+
+    //Total (avg) bits available per frame: total rate over actual/sent frame rate
+    //units are kbits/frame
+    const WebRtc_UWord16 bitRatePerFrame = static_cast<WebRtc_UWord16>(slice_mtu*bitRate/(parameters->frameRate));
+
+    //Total (avg) number of packets per frame (source and fec):
+    const WebRtc_UWord8 avgTotPackets  = 1 + (WebRtc_UWord8)((float)bitRatePerFrame*1000.0/(float)(8.0*_maxPayloadSize) + 0.5);
+
+    //TODO(marpan): Tune model for FEC Protection.
+    //Better modulation of protection with available bits/frame (or avgTotpackets) using weight factors
+    //FEC Tables include this effect already, but need to tune model off-line
+    float weight1 = 0.5;
+    float weight2 = 0.5;
+    if (avgTotPackets > 4)
+    {
+        weight1 = 1.0;
+        weight2 = 0.;
+    }
+    if (avgTotPackets > 6)
+    {
+        weight1 = 1.5;
+        weight2 = 0.;
+    }
+   //
+
+    //Fec rate parameters: for P and I frame
+    WebRtc_UWord8 codeRateDelta = 0;
+    WebRtc_UWord8 codeRateKey = 0;
+
+
+    //Get index for new table: the FEC protection depends on the (avergare) available bits/frame
+    //the range on the rate index corresponds to rates (bps) from 200k to 8000k, for 30fps
+    WebRtc_UWord8 rateIndexTable = (WebRtc_UWord8) VCM_MAX(VCM_MIN((bitRatePerFrame-ratePar1)/ratePar1,ratePar2),0);
+
+    // Restrict packet loss range to 50 for now%: current tables defined only up to 50%
+    if (packetLoss >= plossMax)
+    {
+        packetLoss = plossMax - 1;
+    }
+    WebRtc_UWord16 indexTable = rateIndexTable * plossMax + packetLoss;
+
+    //check on table index
+    if (indexTable >= maxFecTableSize)
+    {
+        assert("FEC table index too large\n");
+    }
+    //
+
+    //For Key frame: effectively at a higher rate, so we scale/boost the rate index.
+    //the boost factor may depend on several factors: ratio of packet number of I to P frames, how much protection placed on P frames, etc.
+    //default is 2
+    const WebRtc_UWord8 packetFrameDelta = (WebRtc_UWord8)(0.5 + parameters->packetsPerFrame);
+    const WebRtc_UWord8 packetFrameKey = (WebRtc_UWord8) (0.5 + parameters->packetsPerFrameKey);
+    const WebRtc_UWord8 boostKey = BoostCodeRateKey(packetFrameDelta, packetFrameKey);
+    rateIndexTable = (WebRtc_UWord8) VCM_MAX(VCM_MIN(1+(boostKey*bitRatePerFrame-ratePar1)/ratePar1,ratePar2),0);
+    WebRtc_UWord16 indexTableKey = rateIndexTable * plossMax + packetLoss;
+
+    indexTableKey = VCM_MIN(indexTableKey, maxFecTableSize);
+
+    codeRateDelta = VCMCodeRateXORTable[indexTable];  //protection factor for P fra
+    codeRateKey = VCMCodeRateXORTable[indexTableKey]; //protection factor for I frame
+
+    //average with minimum protection level given by (average) total number of packets
+    if (packetLoss > 0)
+    {
+        codeRateDelta = static_cast<WebRtc_UWord8>((weight1*(float)codeRateDelta + weight2*255.0/(float)avgTotPackets));
+    }
+
+    //check limit on amount of protection for P frame; 50% is max
+    if (codeRateDelta >= plossMax)
+    {
+        codeRateDelta = plossMax - 1;
+    }
+
+    //make sure I frame protection is at least larger than P frame protection, and at least as high as received loss
+    codeRateKey = static_cast<WebRtc_UWord8>(VCM_MAX(packetLoss,VCM_MAX(_scaleProtKey*codeRateDelta, codeRateKey)));
+
+    //check limit on amount of protection for I frame: 50% is max
+    if (codeRateKey >= plossMax)
+    {
+        codeRateKey = plossMax - 1;
+    }
+
+    _protectionFactorK = codeRateKey;
+    _protectionFactorD = codeRateDelta;
+
+    // DONE WITH FEC PROTECTION SETTINGS
+
+
+    return true;
+}
+
+
+bool
+VCMFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
+{
+
+    // ER SETTINGS:
+    //Effective packet loss to encoder is based on RPL (residual packet loss)
+    //this is a soft setting based on degree of FEC protection
+    //RPL = received/input packet loss - average_FEC_recovery
+    //note: received/input packet loss may be filtered according to FilteredLoss
+
+    //The input packet loss:
+    WebRtc_UWord8 effPacketLoss = (WebRtc_UWord8)(255*parameters->lossPr);
+
+    float scaleErRS = 0.5;
+    float scaleErXOR = 0.5;
+    float minErLevel = (float) 0.025;
+    //float scaleErRS = 1.0;
+    //float scaleErXOR = 1.0;
+    //float minErLevel = (float) 0.0;
+
+    float avgFecRecov = 0.;
+    //Effective packet loss for ER:
+    float scaleEr = scaleErXOR;
+    avgFecRecov = AvgRecoveryFEC(parameters);
+
+    //Residual Packet Loss:
+    _residualPacketLoss = (float)(effPacketLoss - avgFecRecov)/(float)255.0;
+
+
+    //Effective Packet Loss for encoder:
+    _effectivePacketLoss = 0;
+    if (effPacketLoss > 0)
+    {
+        _effectivePacketLoss = VCM_MAX((effPacketLoss - (WebRtc_UWord8)(scaleEr*avgFecRecov)),static_cast<WebRtc_UWord8>(minErLevel*255));
+    }
+
+
+    // DONE WITH ER SETTING
+
+     return true;
+}
+
+
+bool
+VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+
+    // Compute the protection factor
+    ProtectionFactor(parameters);
+
+    // Compute the effective packet loss
+    EffectivePacketLoss(parameters);
+
+
+    // Compute the bit cost
+    // Ignore key frames for now.
+    float fecRate = static_cast<float>(_protectionFactorD) / 255.0f;
+    if (fecRate >= 0.0f)
+    {
+        // use this formula if the fecRate (protection factor) is defined relative to number of source packets
+        // this is the case for the previous tables:
+        // _efficiency = parameters->bitRate * ( 1.0 - 1.0 / (1.0 + fecRate));
+
+        // in the new tables, the fecRate is defined relative to total number of packets (total rate),
+        // so overhead cost is:
+        _efficiency = parameters->bitRate * fecRate;
+    }
+    else
+    {
+        _efficiency = 0.0f;
+    }
+    _score = _efficiency;
+
+
+    // Protection/fec rates obtained above is defined relative to total number of packets (total rate: source+fec)
+    // FEC in RTP module assumes protection factor is defined relative to source number of packets
+    // so we should convert the factor to reduce mismatch between mediaOpt suggested rate and the actual rate
+    _protectionFactorK = ConvertFECRate(_protectionFactorK);
+    _protectionFactorD = ConvertFECRate(_protectionFactorD);
+
+    return true;
+}
+
+bool
+VCMIntraReqMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+    float packetRate = parameters->packetsPerFrame * parameters->frameRate;
+    // Assume that all lost packets cohere to different frames
+    float lossRate = parameters->lossPr * packetRate;
+    if (parameters->keyFrameSize <= 1e-3)
+    {
+        _score = FLT_MAX;
+        return false;
+    }
+    _efficiency = lossRate * parameters->keyFrameSize;
+    _score = _efficiency;
+    if (parameters->lossPr >= 1.0f / parameters->keyFrameSize || parameters->rtt > _IREQ_MAX_RTT)
+    {
+        return false;
+    }
+    return true;
+}
+
+bool
+VCMPeriodicIntraMethod::UpdateParameters(const VCMProtectionParameters* /*parameters*/)
+{
+    // Periodic I-frames. The last thing we want to use.
+    _efficiency = 0.0f;
+    _score = FLT_MAX;
+    return true;
+}
+
+bool
+VCMMbIntraRefreshMethod::UpdateParameters(const VCMProtectionParameters* parameters)
+{
+    // Assume optimal for now.
+    _efficiency = parameters->bitRate * parameters->lossPr / (1.0f + parameters->lossPr);
+    _score = _efficiency;
+    if (parameters->bitRate < _MBREF_MIN_BITRATE)
+    {
+        return false;
+    }
+    return true;
+}
+
+WebRtc_UWord16
+VCMNackMethod::MaxRttNack() const
+{
+    return _NACK_MAX_RTT;
+}
+
+VCMLossProtectionLogic::~VCMLossProtectionLogic()
+{
+    ClearLossProtections();
+}
+
+void
+VCMLossProtectionLogic::ClearLossProtections()
+{
+    ListItem *item;
+    while ((item = _availableMethods.First()) != 0)
+    {
+        VCMProtectionMethod *method = static_cast<VCMProtectionMethod*>(item->GetItem());
+        if (method != NULL)
+        {
+            delete method;
+        }
+        _availableMethods.PopFront();
+    }
+    _selectedMethod = NULL;
+}
+
+ bool
+VCMLossProtectionLogic::AddMethod(VCMProtectionMethod *newMethod)
+{
+    VCMProtectionMethod *method;
+    ListItem *item;
+    if (newMethod == NULL)
+    {
+        return false;
+    }
+    for (item = _availableMethods.First(); item != NULL; item = _availableMethods.Next(item))
+    {
+        method = static_cast<VCMProtectionMethod *>(item->GetItem());
+        if (method != NULL && method->Type() == newMethod->Type())
+        {
+            return false;
+        }
+    }
+    _availableMethods.PushBack(newMethod);
+    return true;
+
+}
+bool
+VCMLossProtectionLogic::RemoveMethod(VCMProtectionMethodEnum methodType)
+{
+    VCMProtectionMethod *method;
+    ListItem *item;
+    bool foundAndRemoved = false;
+    for (item = _availableMethods.First(); item != NULL; item = _availableMethods.Next(item))
+    {
+        method = static_cast<VCMProtectionMethod *>(item->GetItem());
+        if (method != NULL && method->Type() == methodType)
+        {
+            if (_selectedMethod != NULL && _selectedMethod->Type() == method->Type())
+            {
+                _selectedMethod = NULL;
+            }
+            _availableMethods.Erase(item);
+            item = NULL;
+            delete method;
+            foundAndRemoved = true;
+        }
+    }
+    return foundAndRemoved;
+}
+
+VCMProtectionMethod*
+VCMLossProtectionLogic::FindMethod(VCMProtectionMethodEnum methodType) const
+{
+    VCMProtectionMethod *method;
+    ListItem *item;
+    for (item = _availableMethods.First(); item != NULL; item = _availableMethods.Next(item))
+    {
+        method = static_cast<VCMProtectionMethod *>(item->GetItem());
+        if (method != NULL && method->Type() == methodType)
+        {
+            return method;
+        }
+    }
+    return NULL;
+}
+
+float
+VCMLossProtectionLogic::HighestOverhead() const
+{
+    VCMProtectionMethod *method;
+    ListItem *item;
+    float highestOverhead = 0.0f;
+    for (item = _availableMethods.First(); item != NULL; item = _availableMethods.Next(item))
+    {
+        method = static_cast<VCMProtectionMethod *>(item->GetItem());
+        if (method != NULL && method->RequiredBitRate() > highestOverhead)
+        {
+            highestOverhead = method->RequiredBitRate();
+        }
+    }
+    return highestOverhead;
+}
+
+void
+VCMLossProtectionLogic::UpdateRtt(WebRtc_UWord32 rtt)
+{
+    _rtt = rtt;
+}
+
+void
+VCMLossProtectionLogic::UpdateResidualPacketLoss(float residualPacketLoss)
+{
+    _residualPacketLoss = residualPacketLoss;
+}
+
+void
+VCMLossProtectionLogic::UpdateFecType(VCMFecTypes fecType)
+{
+    _fecType = fecType;
+}
+
+void
+VCMLossProtectionLogic::UpdateLossPr(WebRtc_UWord8 lossPr255)
+{
+    WebRtc_UWord32 now = static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp());
+    UpdateMaxLossHistory(lossPr255, now);
+    _lossPr255.Apply(static_cast<float>(now - _lastPrUpdateT), static_cast<float>(lossPr255));
+    _lastPrUpdateT = now;
+    _lossPr = _lossPr255.Value() / 255.0f;
+}
+
+void
+VCMLossProtectionLogic::UpdateMaxLossHistory(WebRtc_UWord8 lossPr255, WebRtc_Word64 now)
+{
+    if (_lossPrHistory[0].timeMs >= 0 &&
+        now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs)
+    {
+        if (lossPr255 > _shortMaxLossPr255)
+        {
+            _shortMaxLossPr255 = lossPr255;
+        }
+    }
+    else
+    {
+        // Only add a new value to the history once a second
+        if(_lossPrHistory[0].timeMs == -1)
+        {
+            // First, no shift
+            _shortMaxLossPr255 = lossPr255;
+        }
+        else
+        {
+            // Shift
+            for(WebRtc_Word32 i = (kLossPrHistorySize - 2); i >= 0 ; i--)
+            {
+                _lossPrHistory[i+1].lossPr255 = _lossPrHistory[i].lossPr255;
+                _lossPrHistory[i+1].timeMs = _lossPrHistory[i].timeMs;
+            }
+        }
+        if (_shortMaxLossPr255 == 0)
+        {
+            _shortMaxLossPr255 = lossPr255;
+        }
+
+         _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
+        _lossPrHistory[0].timeMs = now;
+        _shortMaxLossPr255 = 0;
+
+    }
+}
+
+WebRtc_UWord8
+VCMLossProtectionLogic::MaxFilteredLossPr(WebRtc_Word64 nowMs) const
+{
+    WebRtc_UWord8 maxFound = _shortMaxLossPr255;
+    if (_lossPrHistory[0].timeMs == -1)
+    {
+        return maxFound;
+    }
+    for (WebRtc_Word32 i=0; i < kLossPrHistorySize; i++)
+    {
+        if (_lossPrHistory[i].timeMs == -1)
+        {
+            break;
+        }
+        if (nowMs - _lossPrHistory[i].timeMs > kLossPrHistorySize * kLossPrShortFilterWinMs)
+        {
+            // This sample (and all samples after this) is too old
+            break;
+        }
+        if (_lossPrHistory[i].lossPr255 > maxFound)
+        {
+            // This sample is the largest one this far into the history
+            maxFound = _lossPrHistory[i].lossPr255;
+        }
+    }
+    return maxFound;
+}
+
+WebRtc_UWord8
+VCMLossProtectionLogic::FilteredLoss() const
+{
+
+    //take the average received loss
+    //return static_cast<WebRtc_UWord8>(_lossPr255.Value() + 0.5f);
+
+    //take the windowed max of the received loss
+    if (_selectedMethod != NULL && _selectedMethod->Type() == kFEC)
+    {
+        return MaxFilteredLossPr(static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp()));
+    }
+    else
+    {
+        return static_cast<WebRtc_UWord8>(_lossPr255.Value() + 0.5);
+    }
+
+}
+
+void
+VCMLossProtectionLogic::UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc)
+{
+   _lossPr = (float)packetLossEnc/(float)255.0;
+}
+
+void
+VCMLossProtectionLogic::UpdateBitRate(float bitRate)
+{
+    _bitRate = bitRate;
+}
+
+void
+VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets)
+{
+    WebRtc_UWord32 now = static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp());
+    _packetsPerFrame.Apply(static_cast<float>(now - _lastPacketPerFrameUpdateT), nPackets);
+    _lastPacketPerFrameUpdateT = now;
+}
+
+void
+VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets)
+{
+    WebRtc_UWord32 now = static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp());
+    _packetsPerFrameKey.Apply(static_cast<float>(now - _lastPacketPerFrameUpdateTKey), nPackets);
+    _lastPacketPerFrameUpdateTKey = now;
+}
+
+void
+VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize)
+{
+    _keyFrameSize = keyFrameSize;
+}
+
+bool
+VCMLossProtectionLogic::UpdateMethod(VCMProtectionMethod *newMethod /*=NULL */)
+{
+    _currentParameters.rtt     = _rtt;
+    _currentParameters.lossPr  = _lossPr;
+    _currentParameters.bitRate = _bitRate;
+    _currentParameters.frameRate    = _frameRate; //should this be named actual frame rate?
+    _currentParameters.keyFrameSize = _keyFrameSize;
+    _currentParameters.fecRateDelta = _fecRateDelta;
+    _currentParameters.fecRateKey   = _fecRateKey;
+    _currentParameters.packetsPerFrame = _packetsPerFrame.Value();
+    _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.Value();
+    _currentParameters.residualPacketLoss = _residualPacketLoss;
+    _currentParameters.fecType = _fecType;
+
+    if (newMethod == NULL)
+    {
+        //_selectedMethod = _bestNotOkMethod = NULL;
+        VCMProtectionMethod *method;
+        ListItem *item;
+        for (item = _availableMethods.First(); item != NULL; item = _availableMethods.Next(item))
+        {
+            method = static_cast<VCMProtectionMethod *>(item->GetItem());
+            if (method != NULL)
+            {
+                if (method->Type() == kFEC)
+                {
+                    _selectedMethod = method;
+                }
+                method->UpdateParameters(&_currentParameters);
+            }
+        }
+        if (_selectedMethod != NULL && _selectedMethod->Type() != kFEC)
+        {
+            _selectedMethod = method;
+        }
+    }
+    else
+    {
+        _selectedMethod = newMethod;
+        _selectedMethod->UpdateParameters(&_currentParameters);
+    }
+    return true;
+}
+
+VCMProtectionMethod*
+VCMLossProtectionLogic::SelectedMethod() const
+{
+    return _selectedMethod;
+}
+
+void
+VCMLossProtectionLogic::Reset()
+{
+    _lastPrUpdateT = static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp());
+    _lastPacketPerFrameUpdateT = static_cast<WebRtc_UWord32>(VCMTickTime::MillisecondTimestamp());
+    _lossPr255.Reset(0.9999f);
+    _packetsPerFrame.Reset(0.9999f);
+    _fecRateDelta = _fecRateKey = 0;
+    for (WebRtc_Word32 i=0; i < kLossPrHistorySize; i++)
+    {
+        _lossPrHistory[i].lossPr255 = 0;
+        _lossPrHistory[i].timeMs = -1;
+    }
+    _shortMaxLossPr255 = 0;
+    ClearLossProtections();
+}
+
+}

modules/video_coding/main/source/media_opt_util.h

@@ -0,0 +1,372 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+
+#include "typedefs.h"
+#include "list_wrapper.h"
+#include "trace.h"
+#include "exp_filter.h"
+#include "internal_defines.h"
+#include "tick_time.h"
+
+#include <cmath>
+#include <cstdlib>
+
+
+namespace webrtc
+{
+class ListWrapper;
+
+enum { kLossPrHistorySize = 30 }; // 30 time periods
+enum { kLossPrShortFilterWinMs = 1000 }; // 1000 ms, total filter length is 30 000 ms
+
+enum VCMFecTypes
+{
+    kXORFec
+};
+
+struct VCMProtectionParameters
+{
+    VCMProtectionParameters() : rtt(0), lossPr(0), bitRate(0), packetsPerFrame(0),
+        frameRate(0), keyFrameSize(0), fecRateDelta(0), fecRateKey(0),
+        residualPacketLoss(0.0), fecType(kXORFec) {}
+
+    WebRtc_UWord32      rtt;
+    float               lossPr;
+    float               bitRate;
+    float               packetsPerFrame;
+    float               packetsPerFrameKey;
+    float               frameRate;
+    float               keyFrameSize;
+    WebRtc_UWord8       fecRateDelta;
+    WebRtc_UWord8       fecRateKey;
+    float               residualPacketLoss;
+    VCMFecTypes           fecType;
+};
+
+
+/******************************/
+/* VCMProtectionMethod class    */
+/****************************/
+
+enum VCMProtectionMethodEnum
+{
+    kNACK,
+    kFEC,
+    kNackFec,
+    kIntraRequest, // I-frame request
+    kPeriodicIntra, // I-frame refresh
+    kMBIntraRefresh, // Macro block refresh
+    kNone
+};
+
+class VCMLossProbabilitySample
+{
+public:
+    VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {};
+
+    WebRtc_UWord8     lossPr255;
+    WebRtc_Word64     timeMs;
+};
+
+
+
+class VCMProtectionMethod
+{
+public:
+    //friend VCMProtectionMethod;
+    VCMProtectionMethod(VCMProtectionMethodEnum type) : _protectionFactorK(0),
+        _protectionFactorD(0), _residualPacketLoss(0.0), _scaleProtKey(2.0),
+        _maxPayloadSize(1460), _efficiency(0), _score(0), _type(type) {}
+    virtual ~VCMProtectionMethod() {}
+
+    // Updates the efficiency of the method using the parameters provided
+    //
+    // Input:
+    //         - parameters         : Parameters used to calculate the efficiency
+    //
+    // Return value                 : True if this method is recommended in
+    //                                the given conditions.
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
+
+    // Returns the protection type
+    //
+    // Return value                 : The protection type
+    enum VCMProtectionMethodEnum Type() const { return _type; }
+
+    // Evaluates if this protection method is considered
+    // better than the provided method.
+    //
+    // Input:
+    //          - pm                : The protection method to compare with
+    bool BetterThan(VCMProtectionMethod *pm);
+
+    // Returns the bit rate required by this protection method
+    // during these conditions.
+    //
+    // Return value                 : Required bit rate
+    virtual float RequiredBitRate() { return _efficiency; }
+
+    // Returns the effective packet loss for ER, required by this protection method
+    //
+    // Return value                 : Required effective packet loss
+    virtual WebRtc_UWord8 RequiredPacketLossER() { return _effectivePacketLoss; }
+
+    // Extracts the FEC protection factor for Key frame, required by this protection method
+    //
+    // Return value                 : Required protectionFactor for Key frame
+    virtual WebRtc_UWord8 RequiredProtectionFactorK() { return _protectionFactorK; }
+
+    // Extracts the FEC protection factor for Delta frame, required by this protection method
+    //
+    // Return value                 : Required protectionFactor for delta frame
+    virtual WebRtc_UWord8 RequiredProtectionFactorD() { return _protectionFactorD; }
+
+    WebRtc_UWord8                        _effectivePacketLoss;
+    WebRtc_UWord8                        _protectionFactorK;
+    WebRtc_UWord8                        _protectionFactorD;
+    float                              _residualPacketLoss;
+    float                              _scaleProtKey;
+    WebRtc_Word32                        _maxPayloadSize;
+
+protected:
+    float                              _efficiency;
+    float                              _score;
+
+private:
+    const enum VCMProtectionMethodEnum     _type;
+
+};
+
+class VCMNackMethod : public VCMProtectionMethod
+{
+public:
+    VCMNackMethod() : VCMProtectionMethod(kNACK), _NACK_MAX_RTT(200) {}
+    virtual ~VCMNackMethod() {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+    //get the effective packet loss for ER
+    bool EffectivePacketLoss(WebRtc_UWord8 effPacketLoss, WebRtc_UWord16 rttTime);
+    //get the threshold for NACK
+    WebRtc_UWord16 MaxRttNack() const;
+private:
+    const WebRtc_UWord16 _NACK_MAX_RTT;
+};
+
+class VCMFecMethod : public VCMProtectionMethod
+{
+public:
+    VCMFecMethod() : VCMProtectionMethod(kFEC) {}
+    virtual ~VCMFecMethod() {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+    //get the effective packet loss for ER
+    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+    //get the FEC protection factors
+    bool ProtectionFactor(const VCMProtectionParameters* parameters);
+    //get the boost for key frame protection
+    WebRtc_UWord8 BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta,
+                                   WebRtc_UWord8 packetFrameKey) const;
+    //convert the rates: defined relative to total# packets or source# packets
+    WebRtc_UWord8 ConvertFECRate(WebRtc_UWord8 codeRate) const;
+    //get the average effective recovery from FEC: for random loss model
+    float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
+};
+
+
+class VCMNackFecMethod : public VCMProtectionMethod
+{
+public:
+    VCMNackFecMethod() : VCMProtectionMethod(kNackFec) {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+    //get the effective packet loss for ER
+    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+    //get the FEC protection factors
+    bool ProtectionFactor(const VCMProtectionParameters* parameters);
+
+};
+
+
+class VCMIntraReqMethod : public VCMProtectionMethod
+{
+public:
+    VCMIntraReqMethod() : VCMProtectionMethod(kIntraRequest), _IREQ_MAX_RTT(150) {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+private:
+    const WebRtc_UWord32 _IREQ_MAX_RTT;
+};
+
+class VCMPeriodicIntraMethod : public VCMProtectionMethod
+{
+public:
+    VCMPeriodicIntraMethod() : VCMProtectionMethod(kPeriodicIntra) {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+};
+
+class VCMMbIntraRefreshMethod : public VCMProtectionMethod
+{
+public:
+    VCMMbIntraRefreshMethod() :
+        VCMProtectionMethod(kMBIntraRefresh), _MBREF_MIN_BITRATE(150) {}
+    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+    virtual float RequiredBitRate() { return 0.0; }
+private:
+    const WebRtc_UWord32 _MBREF_MIN_BITRATE;
+};
+
+class VCMLossProtectionLogic
+{
+public:
+    VCMLossProtectionLogic() : _availableMethods(), _selectedMethod(NULL),
+        _bestNotOkMethod(NULL), _rtt(0), _lossPr(0.0f), _bitRate(0.0f), _frameRate(0.0f),
+        _keyFrameSize(0.0f), _fecRateKey(0), _fecRateDelta(0), _lastPrUpdateT(0),
+        _lossPr255(0.9999f), _lossPrHistory(), _shortMaxLossPr255(0),
+        _packetsPerFrame(0.9999f), _packetsPerFrameKey(0.9999f), _residualPacketLoss(0),
+        _boostRateKey(2)
+    { Reset(); }
+
+    ~VCMLossProtectionLogic();
+
+    void ClearLossProtections();
+    bool AddMethod(VCMProtectionMethod *newMethod);
+    bool RemoveMethod(VCMProtectionMethodEnum methodType);
+    VCMProtectionMethod* FindMethod(VCMProtectionMethodEnum methodType) const;
+    float HighestOverhead() const;
+
+    // Update the round-trip time
+    //
+    // Input:
+    //          - rtt           : Round-trip time in seconds.
+    void UpdateRtt(WebRtc_UWord32 rtt);
+
+    // Update residual packet loss
+    //
+    // Input:
+    //          - residualPacketLoss  : residual packet loss: effective loss after FEC recovery
+    void UpdateResidualPacketLoss(float residualPacketLoss);
+
+    // Update fecType
+    //
+    // Input:
+    //          - fecType           : kXORFec for generic XOR FEC
+    void UpdateFecType(VCMFecTypes fecType);
+
+    // Update the loss probability.
+    //
+    // Input:
+    //          - lossPr255        : The packet loss probability in the interval [0, 255],
+    //                               reported by RTCP.
+    void UpdateLossPr(WebRtc_UWord8 lossPr255);
+
+    // Update the filtered packet loss.
+    //
+    // Input:
+    //          - packetLossEnc :  The reported packet loss filtered (max window or average)
+    void UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc);
+
+    // Update the current target bit rate.
+    //
+    // Input:
+    //          - bitRate          : The current target bit rate in kbits/s
+    void UpdateBitRate(float bitRate);
+
+    // Update the number of packets per frame estimate, for delta frames
+    //
+    // Input:
+    //          - nPackets         : Number of packets used to send the latest frame.
+    void UpdatePacketsPerFrame(float nPackets);
+
+   // Update the number of packets per frame estimate, for key frames
+    //
+    // Input:
+    //          - nPackets         : Number of packets used to send the latest frame.
+    void UpdatePacketsPerFrameKey(float nPackets);
+
+    // Update the keyFrameSize estimate
+    //
+    // Input:
+    //          - keyFrameSize     : The size of the latest sent key frame.
+    void UpdateKeyFrameSize(float keyFrameSize);
+
+    // Update the frame rate
+    //
+    // Input:
+    //          - frameRate        : The current target frame rate.
+    void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
+
+    // The amount of packet loss to cover for with FEC.
+    //
+    // Input:
+    //          - fecRateKey      : Packet loss to cover for with FEC when sending key frames.
+    //          - fecRateDelta    : Packet loss to cover for with FEC when sending delta frames.
+    void UpdateFECRates(WebRtc_UWord8 fecRateKey, WebRtc_UWord8 fecRateDelta)
+                       { _fecRateKey = fecRateKey; _fecRateDelta = fecRateDelta; }
+
+    // Update the protection methods with the current VCMProtectionParameters and
+    // choose the best method available. The update involves computing the robustness settings
+    // for the protection method.
+    //
+    // Input:
+    //          - newMethod         : If not NULL, this is method will be selected by force.
+    //
+    // Return value     : True if the selected method is recommended using these settings,
+    //                    false if it's the best method, but still not recommended to be used.
+    //                    E.g. if NACK is the best available, but the RTT is too large, false
+    //                    will be returned.
+    bool UpdateMethod(VCMProtectionMethod *newMethod = NULL);
+
+    // Returns the method currently selected.
+    //
+    // Return value                 : The protection method currently selected.
+    VCMProtectionMethod* SelectedMethod() const;
+
+    // Returns the filtered loss probability in the interval [0, 255].
+    //
+    // Return value                 : The filtered loss probability
+    WebRtc_UWord8 FilteredLoss() const;
+
+    // Get constraint on NACK
+    //
+    // return value                : RTT threshold for using NACK
+   WebRtc_UWord8  GetNackThreshold() const;
+
+    void Reset();
+
+private:
+    // Sets the available loss protection methods.
+    void UpdateMaxLossHistory(WebRtc_UWord8 lossPr255, WebRtc_Word64 now);
+    WebRtc_UWord8 MaxFilteredLossPr(WebRtc_Word64 nowMs) const;
+    ListWrapper                  _availableMethods;
+    VCMProtectionMethod*      _selectedMethod;
+    VCMProtectionMethod*      _bestNotOkMethod;
+    VCMProtectionParameters   _currentParameters;
+    WebRtc_UWord32            _rtt;
+    float                     _lossPr;
+    float                     _bitRate;
+    float                     _frameRate;
+    float                     _keyFrameSize;
+    WebRtc_UWord8             _fecRateKey;
+    WebRtc_UWord8             _fecRateDelta;
+    WebRtc_UWord32            _lastPrUpdateT;
+    WebRtc_UWord32            _lastPacketPerFrameUpdateT;
+    WebRtc_UWord32            _lastPacketPerFrameUpdateTKey;
+    VCMExpFilter              _lossPr255;
+    VCMLossProbabilitySample  _lossPrHistory[kLossPrHistorySize];
+    WebRtc_UWord8             _shortMaxLossPr255;
+    VCMExpFilter              _packetsPerFrame;
+    VCMExpFilter              _packetsPerFrameKey;
+    float                     _residualPacketLoss;
+    WebRtc_UWord8             _boostRateKey;
+    VCMFecTypes               _fecType;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_

modules/video_coding/main/source/media_optimization.cc

@@ -0,0 +1,687 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "media_optimization.h"
+#include "content_metrics_processing.h"
+#include "frame_dropper.h"
+#include "qm_select.h"
+
+namespace webrtc {
+
+VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id):
+_id(id),
+_maxBitRate(0),
+_sendCodecType(kVideoCodecUnknown),
+_codecWidth(0),
+_codecHeight(0),
+_userFrameRate(0),
+_lossProtOverhead(0),
+_packetLossEnc(0),
+_fractionLost(0),
+_sendStatisticsZeroEncode(0),
+_maxPayloadSize(1460),
+_lastBitRate(0),
+_targetBitRate(0),
+_enableQm(false),
+_videoProtectionCallback(NULL),
+_videoQMSettingsCallback(NULL),
+_encodedFrameSamples(),
+_avgSentBitRateBps(0.0f),
+_keyFrameCnt(0),
+_deltaFrameCnt(0),
+_lastQMUpdateTime(0),
+_lastChangeTime(0)
+{
+    memset(_sendStatistics, 0, sizeof(_sendStatistics));
+
+    _frameDropper  = new VCMFrameDropper(_id);
+    _lossProtLogic = new VCMLossProtectionLogic();
+    _content = new VCMContentMetricsProcessing();
+    _qms = new VCMQmSelect();
+}
+
+VCMMediaOptimization::~VCMMediaOptimization(void)
+{
+    _lossProtLogic->ClearLossProtections();
+    delete _lossProtLogic;
+    delete _frameDropper;
+    delete _content;
+    delete _qms;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::Reset()
+{
+    _frameDropper->Reset();
+    _lossProtLogic->Reset();
+    _frameDropper->SetRates(0, 0);
+    _content->Reset();
+    _qms->Reset();
+    _lossProtLogic->UpdateFrameRate(static_cast<float>(InputFrameRate()));
+    _lossProtLogic->Reset();
+    _sendStatisticsZeroEncode = 0;
+    _lastBitRate = 0;
+    _targetBitRate = 0;
+    _lossProtOverhead = 0;
+    _codecWidth = 0;
+    _codecHeight = 0;
+    _userFrameRate = 0;
+    _keyFrameCnt = 0;
+    _deltaFrameCnt = 0;
+    _lastQMUpdateTime = 0;
+    _lastChangeTime = 0;
+    for (WebRtc_Word32 i = 0; i < kBitrateMaxFrameSamples; i++)
+    {
+        _encodedFrameSamples[i]._sizeBytes = -1;
+        _encodedFrameSamples[i]._timeCompleteMs = -1;
+    }
+    _avgSentBitRateBps = 0.0f;
+    return VCM_OK;
+}
+
+WebRtc_UWord32
+VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
+                                     WebRtc_UWord8 &fractionLost,
+                                     WebRtc_UWord32 roundTripTimeMs)
+{
+    VCMProtectionMethod *selectedMethod = _lossProtLogic->SelectedMethod();
+    _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
+    _lossProtLogic->UpdateLossPr(fractionLost);
+    _lossProtLogic->UpdateRtt(roundTripTimeMs);
+    _lossProtLogic->UpdateResidualPacketLoss(static_cast<float>(fractionLost));
+
+    VCMFecTypes fecType = kXORFec;  // generic FEC
+    _lossProtLogic->UpdateFecType(fecType);
+
+    //Get frame rate for encoder: this is the actual/sent frame rate
+    float actualFrameRate = SentFrameRate();
+
+    // sanity
+    if (actualFrameRate  < 1.0)
+    {
+        actualFrameRate = 1.0;
+    }
+
+    // Update frame rate for the loss protection logic class: frame rate should be the actual/sent rate
+    _lossProtLogic->UpdateFrameRate(actualFrameRate);
+
+    _fractionLost = fractionLost;
+
+    // The effective packet loss may be the received loss or filtered, i.e., average or max filter may be used.
+    //We should think about which filter is appropriate for low/high bit rates, low/high loss rates, etc.
+    WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss();
+
+    //For now use the filtered loss for computing the robustness settings
+    _lossProtLogic->UpdateFilteredLossPr(packetLossEnc);
+
+    // Rate cost of the protection methods
+    _lossProtOverhead = 0;
+
+    if(selectedMethod)
+    {
+
+        //Update method will compute the robustness settings for the given protection method and the overhead cost
+        //the protection method is set by the user via SetVideoProtection.
+        //The robustness settings are: the effecitve packet loss for ER and the FEC protection settings
+        _lossProtLogic->UpdateMethod();
+
+        //Get the code rate for Key frames
+        const WebRtc_UWord8 codeRateKeyRTP  = selectedMethod->RequiredProtectionFactorK();
+
+        //Get the code rate for Delta frames
+        const WebRtc_UWord8 codeRateDeltaRTP = selectedMethod->RequiredProtectionFactorD();
+
+        //Get the effective packet loss for ER
+        packetLossEnc = selectedMethod->RequiredPacketLossER();
+
+        // Get the bit cost of protection method
+        _lossProtOverhead = static_cast<WebRtc_UWord32>(_lossProtLogic->HighestOverhead() + 0.5f);
+
+        //NACK is on for NACK and NackFec protection method: off for FEC method
+        bool nackStatus = true;
+        if (selectedMethod->Type() == kFEC)
+        {
+            nackStatus = false;
+        }
+
+        if(_videoProtectionCallback)
+        {
+            _videoProtectionCallback->ProtectionRequest(codeRateDeltaRTP ,codeRateKeyRTP, nackStatus);
+        }
+
+    }
+
+    // Update effective packet loss for encoder: note: fractionLost was passed as reference
+    fractionLost = packetLossEnc;
+
+    WebRtc_UWord32 nackBitRate=0;
+    if(selectedMethod && _lossProtLogic->FindMethod(kNACK) != NULL)
+    {
+        // Make sure we don't over-use the channel momentarily. This is
+        // necessary for NACK since it can be very bursty.
+        nackBitRate = (_lastBitRate * fractionLost) / 255;
+        if (nackBitRate > _targetBitRate)
+        {
+            nackBitRate = _targetBitRate;
+        }
+        _frameDropper->SetRates(static_cast<float>(bitRate - nackBitRate), 0);
+    }
+    else
+    {
+        _frameDropper->SetRates(static_cast<float>(bitRate - _lossProtOverhead), 0);
+    }
+
+    //This may be used for UpdateEncoderBitRate: lastBitRate is total rate, before compensation
+    _lastBitRate = _targetBitRate;
+
+    //Source coding rate: total rate - protection overhead
+    _targetBitRate = bitRate - _lossProtOverhead;
+
+    if (_enableQm)
+    {
+        //Update QM with rates
+        _qms->UpdateRates((float)_targetBitRate, _avgSentBitRateBps,_incomingFrameRate);
+        //Check for QM selection
+        bool selectQM = checkStatusForQMchange();
+        if (selectQM)
+        {
+            SelectQuality();
+        }
+    }
+
+    return _targetBitRate;
+}
+
+
+bool
+VCMMediaOptimization::DropFrame()
+{
+    _frameDropper->Leak((WebRtc_UWord32)(InputFrameRate() + 0.5f)); // leak appropriate number of bytes
+    return _frameDropper->DropFrame();
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SentFrameCount(VCMFrameCount &frameCount) const
+{
+    frameCount.numDeltaFrames = _deltaFrameCnt;
+    frameCount.numKeyFrames = _keyFrameCnt;
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SetEncodingData(VideoCodecType sendCodecType, WebRtc_Word32 maxBitRate,
+                                      WebRtc_UWord32 frameRate, WebRtc_UWord32 bitRate,
+                                      WebRtc_UWord16 width, WebRtc_UWord16 height)
+{
+    // Everything codec specific should be reset here since this means the codec has changed.
+    // If native dimension values have changed, then either user initiated change, or QM
+    // initiated change. Will be able to determine only after the processing of the first frame
+    _lastChangeTime = VCMTickTime::MillisecondTimestamp();
+    _content->Reset();
+    _content->UpdateFrameRate(frameRate);
+
+    _maxBitRate = maxBitRate;
+    _sendCodecType = sendCodecType;
+    _targetBitRate = bitRate;
+    _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
+    _lossProtLogic->UpdateFrameRate(static_cast<float>(frameRate));
+    _frameDropper->Reset();
+    _frameDropper->SetRates(static_cast<float>(bitRate), static_cast<float>(frameRate));
+    _userFrameRate = (float)frameRate;
+    _codecWidth = width;
+    _codecHeight = height;
+    WebRtc_Word32 ret = VCM_OK;
+    ret = _qms->Initialize((float)_targetBitRate, _userFrameRate, _codecWidth, _codecHeight);
+    return ret;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::RegisterProtectionCallback(VCMProtectionCallback* protectionCallback)
+{
+    _videoProtectionCallback = protectionCallback;
+    return VCM_OK;
+
+}
+
+
+void
+VCMMediaOptimization::EnableFrameDropper(bool enable)
+{
+    _frameDropper->Enable(enable);
+}
+
+
+void
+VCMMediaOptimization::EnableNack(bool enable)
+{
+    // Add NACK to the list of loss protection methods
+    bool updated = false;
+    if (enable)
+    {
+        VCMProtectionMethod *nackMethod = new VCMNackMethod();
+        updated = _lossProtLogic->AddMethod(nackMethod);
+        if (!updated)
+        {
+            delete nackMethod;
+        }
+    }
+    else
+    {
+        updated = _lossProtLogic->RemoveMethod(kNACK);
+    }
+    if (updated)
+    {
+        _lossProtLogic->UpdateMethod();
+    }
+}
+
+bool
+VCMMediaOptimization::IsNackEnabled()
+{
+    return (_lossProtLogic->FindMethod(kFEC) != NULL);
+}
+
+void
+VCMMediaOptimization::EnableFEC(bool enable)
+{
+    // Add FEC to the list of loss protection methods
+    bool updated = false;
+    if (enable)
+    {
+        VCMProtectionMethod *fecMethod = new VCMFecMethod();
+        updated = _lossProtLogic->AddMethod(fecMethod);
+        if (!updated)
+        {
+            delete fecMethod;
+        }
+    }
+    else
+    {
+        updated = _lossProtLogic->RemoveMethod(kFEC);
+    }
+    if (updated)
+    {
+        _lossProtLogic->UpdateMethod();
+    }
+}
+void
+VCMMediaOptimization::EnableNackFEC(bool enable)
+{
+    // Add NackFec to the list of loss protection methods
+    bool updated = false;
+    if (enable)
+    {
+        VCMProtectionMethod *nackfecMethod = new VCMNackFecMethod();
+        updated = _lossProtLogic->AddMethod(nackfecMethod);
+        if (!updated)
+        {
+            delete nackfecMethod;
+        }
+    }
+    else
+    {
+        updated = _lossProtLogic->RemoveMethod(kNackFec);
+    }
+    if (updated)
+    {
+        _lossProtLogic->UpdateMethod();
+    }
+}
+
+bool
+VCMMediaOptimization::IsFecEnabled()
+{
+    return (_lossProtLogic->FindMethod(kFEC) != NULL);
+}
+
+bool
+VCMMediaOptimization::IsNackFecEnabled()
+{
+    return (_lossProtLogic->FindMethod(kNackFec) != NULL);
+}
+
+void
+VCMMediaOptimization::SetMtu(WebRtc_Word32 mtu)
+{
+    _maxPayloadSize = mtu;
+}
+
+float
+VCMMediaOptimization::SentFrameRate()
+{
+    if(_frameDropper)
+    {
+        return _frameDropper->ActualFrameRate((WebRtc_UWord32)(InputFrameRate() + 0.5f));
+    }
+
+    return VCM_CODEC_ERROR;
+}
+
+float
+VCMMediaOptimization::SentBitRate()
+{
+    UpdateBitRateEstimate(-1, VCMTickTime::MillisecondTimestamp());
+    return _avgSentBitRateBps / 1000.0f;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::MaxBitRate()
+{
+    return _maxBitRate;
+}
+
+WebRtc_Word32
+VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
+                                            FrameType encodedFrameType)
+{
+    // look into the ViE version - debug mode - needs also number of layers.
+    UpdateBitRateEstimate(encodedLength, VCMTickTime::MillisecondTimestamp());
+    if(encodedLength > 0)
+    {
+        const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
+                                 encodedFrameType != kVideoFrameGolden);
+
+        _frameDropper->Fill(encodedLength, deltaFrame);
+        if (_maxPayloadSize > 0 && encodedLength > 0)
+        {
+            const float minPacketsPerFrame = encodedLength /
+                                             static_cast<float>(_maxPayloadSize);
+            if (deltaFrame)
+            {
+                _lossProtLogic->UpdatePacketsPerFrame(minPacketsPerFrame);
+            }
+            else
+            {
+                _lossProtLogic->UpdatePacketsPerFrameKey(minPacketsPerFrame);
+            }
+
+            if (_enableQm)
+            {
+                // update quality select with encoded length
+                _qms->UpdateEncodedSize(encodedLength, encodedFrameType);
+            }
+        }
+        if (!deltaFrame && encodedLength > 0)
+        {
+            _lossProtLogic->UpdateKeyFrameSize(static_cast<float>(encodedLength));
+        }
+
+        // updating counters
+        if (deltaFrame){
+            _deltaFrameCnt++;
+        } else {
+            _keyFrameCnt++;
+        }
+
+    }
+
+     return VCM_OK;
+
+}
+
+void VCMMediaOptimization::UpdateBitRateEstimate(WebRtc_Word64 encodedLength,
+                                                 WebRtc_Word64 nowMs)
+{
+    int i = kBitrateMaxFrameSamples - 1;
+    WebRtc_UWord32 frameSizeSum = 0;
+    WebRtc_Word64 timeOldest = -1;
+    // Find an empty slot for storing the new sample and at the same time
+    // accumulate the history.
+    for (; i >= 0; i--)
+    {
+        if (_encodedFrameSamples[i]._sizeBytes == -1)
+        {
+            // Found empty slot
+            break;
+        }
+        if (nowMs - _encodedFrameSamples[i]._timeCompleteMs < kBitrateAverageWinMs)
+        {
+            frameSizeSum += static_cast<WebRtc_UWord32>(_encodedFrameSamples[i]._sizeBytes);
+            if (timeOldest == -1)
+            {
+                timeOldest = _encodedFrameSamples[i]._timeCompleteMs;
+            }
+        }
+    }
+    if (encodedLength > 0)
+    {
+        if (i < 0)
+        {
+            // No empty slot, shift
+            for (i = kBitrateMaxFrameSamples - 2; i >= 0; i--)
+            {
+                _encodedFrameSamples[i + 1] = _encodedFrameSamples[i];
+            }
+            i++;
+        }
+        // Insert new sample
+        _encodedFrameSamples[i]._sizeBytes = encodedLength;
+        _encodedFrameSamples[i]._timeCompleteMs = nowMs;
+    }
+    if (timeOldest > -1)
+    {
+        // Update average bit rate
+        float denom = static_cast<float>(nowMs - timeOldest);
+        if (denom < 1.0)
+        {
+            denom = 1.0;
+        }
+        _avgSentBitRateBps = (frameSizeSum + encodedLength) * 8 * 1000 / denom;
+    }
+    else if (encodedLength > 0)
+    {
+        _avgSentBitRateBps = static_cast<float>(encodedLength * 8);
+    }
+    else
+    {
+        _avgSentBitRateBps = 0;
+    }
+}
+
+
+WebRtc_Word32
+VCMMediaOptimization::RegisterVideoQMCallback(VCMQMSettingsCallback *videoQMSettings)
+{
+    _videoQMSettingsCallback = videoQMSettings;
+    //Callback setting controls QM
+    if (_videoQMSettingsCallback != NULL)
+    {
+        _enableQm = true;
+    }
+    else
+    {
+        _enableQm = false;
+    }
+    return VCM_OK;
+}
+
+void
+VCMMediaOptimization::updateContentData(const VideoContentMetrics *contentMetrics)
+{
+    //Updating content metrics
+    if (contentMetrics == NULL)
+    {
+         //No QM if metrics are NULL
+         _enableQm = false;
+         _qms->Reset();
+    }
+    else
+    {
+        _content->UpdateContentData(contentMetrics);
+    }
+}
+
+WebRtc_Word32
+VCMMediaOptimization::SelectQuality()
+{
+    // Reset quantities for QM select
+    _qms->ResetQM();
+
+    // Select quality mode
+    VCMQualityMode* qm = NULL;
+    WebRtc_Word32 ret = _qms->SelectQuality(_content->Data(), &qm);
+    if (ret < 0)
+    {
+          return ret;
+    }
+
+    // Check for updates to spatial/temporal modes
+    QMUpdate(qm);
+
+    //Reset all the rate and related frame counters quantities
+    _qms->ResetRates();
+
+    // Reset counters
+    _lastQMUpdateTime = VCMTickTime::MillisecondTimestamp();
+
+    // Reset content metrics
+    _content->Reset();
+
+    return VCM_OK;
+}
+
+
+// Check timing constraints and look for significant change in:
+// (1) scene content
+// (2) target bit rate
+
+bool
+VCMMediaOptimization::checkStatusForQMchange()
+{
+
+    bool status  = true;
+
+    // Check that we do not call QMSelect too often, and that we waited some time (to sample the metrics) from the event lastChangeTime
+    // lastChangeTime is the time where user changed the size/rate/frame rate (via SetEncodingData)
+    WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
+    if ((now - _lastQMUpdateTime) < kQmMinIntervalMs ||
+        (now  - _lastChangeTime) <  kQmMinIntervalMs)
+    {
+        status = false;
+    }
+
+    return status;
+
+}
+
+bool
+VCMMediaOptimization::QMUpdate(VCMQualityMode* qm)
+{
+    //Check for no change
+    if (qm->spatialHeightFact == 1 &&
+        qm->spatialWidthFact == 1 &&
+        qm->temporalFact == 1)
+    {
+        return false;
+    }
+
+    //Content metrics hold native values
+    VideoContentMetrics* cm = _content->Data();
+
+    //Temporal
+    WebRtc_UWord32 frameRate  = static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
+    //Check if go back up in temporal resolution
+    if (qm->temporalFact == 0)
+    {
+        frameRate = (WebRtc_UWord32) 2 * _incomingFrameRate;
+    }
+    //go down in temporal resolution
+    else
+    {
+        frameRate = (WebRtc_UWord32)(_incomingFrameRate / qm->temporalFact + 1);
+    }
+
+    //Spatial
+    WebRtc_UWord32 height = _codecHeight;
+    WebRtc_UWord32 width = _codecWidth;
+    //Check if go back up in spatial resolution
+    if (qm->spatialHeightFact == 0 && qm->spatialWidthFact == 0)
+    {
+       height = cm->nativeHeight;
+       width = cm->nativeWidth;
+    }
+    else
+    {
+        height = _codecHeight / qm->spatialHeightFact;
+        width = _codecWidth / qm->spatialWidthFact;
+    }
+
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, _id,
+               "Quality Mode Update: W = %d, H = %d, FR = %f",
+               width, height, frameRate);
+
+    //Update VPM with new target frame rate and size
+    _videoQMSettingsCallback->SetVideoQMSettings(frameRate, width, height);
+
+    return true;
+}
+
+
+
+void
+VCMMediaOptimization::UpdateIncomingFrameRate()
+{
+    WebRtc_Word64 now = VCMTickTime::MillisecondTimestamp();
+    if(_incomingFrameTimes[0] == 0)
+    {
+        // first no shift
+    } else
+    {
+        // shift
+        for(WebRtc_Word32 i = (kFrameCountHistorySize - 2); i >= 0 ; i--)
+        {
+            _incomingFrameTimes[i+1] = _incomingFrameTimes[i];
+        }
+    }
+    _incomingFrameTimes[0] = now;
+    ProcessIncomingFrameRate(now);
+}
+
+// allowing VCM to keep track of incoming frame rate
+void
+VCMMediaOptimization::ProcessIncomingFrameRate(WebRtc_Word64 now)
+{
+    WebRtc_Word32 num = 0;
+    WebRtc_Word32 nrOfFrames = 0;
+    for(num = 1; num < (kFrameCountHistorySize - 1); num++)
+    {
+        if (_incomingFrameTimes[num] <= 0 ||
+            // don't use data older than 2 s
+            now - _incomingFrameTimes[num] > kFrameHistoryWinMs)
+        {
+            break;
+        } else
+        {
+            nrOfFrames++;
+        }
+    }
+    if (num > 1)
+    {
+        const WebRtc_Word64 diff = now - _incomingFrameTimes[num-1];
+        _incomingFrameRate = 1.0;
+        if(diff >0)
+        {
+            _incomingFrameRate = nrOfFrames * 1000.0f / static_cast<float>(diff);
+        }
+    }
+    else
+    {
+        _incomingFrameRate = static_cast<float>(nrOfFrames);
+    }
+}
+
+WebRtc_UWord32
+VCMMediaOptimization::InputFrameRate()
+{
+    ProcessIncomingFrameRate(VCMTickTime::MillisecondTimestamp());
+    return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
+}
+
+}

modules/video_coding/main/source/media_optimization.h

@@ -0,0 +1,220 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+
+#include "list_wrapper.h"
+#include "module_common_types.h"
+#include "video_coding.h"
+#include "trace.h"
+#include "media_opt_util.h"
+#include "qm_select.h"
+
+namespace webrtc
+{
+
+enum { kBitrateMaxFrameSamples = 60 };
+enum { kBitrateAverageWinMs    = 1000 };
+
+class VCMContentMetricsProcessing;
+class VCMFrameDropper;
+
+struct VCMEncodedFrameSample
+{
+    VCMEncodedFrameSample() : _sizeBytes(-1), _timeCompleteMs(-1) {}
+
+    WebRtc_Word64     _sizeBytes;
+    WebRtc_Word64     _timeCompleteMs;
+};
+
+class VCMMediaOptimization
+{
+public:
+    VCMMediaOptimization(WebRtc_Word32 id);
+    ~VCMMediaOptimization(void);
+    /*
+    * Reset the Media Optimization module
+    */
+    WebRtc_Word32 Reset();
+    /**
+    * Set target Rates for the encoder given the channel parameters
+    * Inputs:       bitRate - target bitRate, in the conference case this is the rate
+    *                         between the sending client and the server
+    *               fractionLost - packet loss in % in the network
+    *               roundTripTimeMs - round trip time in miliseconds
+    *               minBitRate - the bit rate of the end-point with lowest rate
+    *               maxBitRate - the bit rate of the end-point with highest rate
+    */
+    WebRtc_UWord32 SetTargetRates(WebRtc_UWord32 bitRate,
+                                  WebRtc_UWord8 &fractionLost,
+                                  WebRtc_UWord32 roundTripTimeMs);
+
+    /**
+    * Inform media optimization of initial encoding state
+    */
+    WebRtc_Word32 SetEncodingData(VideoCodecType sendCodecType,
+                                  WebRtc_Word32 maxBitRate,
+                                  WebRtc_UWord32 frameRate,
+                                  WebRtc_UWord32 bitRate,
+                                  WebRtc_UWord16 width,
+                                  WebRtc_UWord16 height);
+    /**
+    * Enable NACK and update error resilience parameters
+    */
+    void EnableNack(bool enable);
+    /**
+    * Returns weather or not NACK is enabled
+    */
+    bool IsNackEnabled();
+    /**
+    * Enable FEC and update error resilience parameters
+    */
+    void EnableFEC(bool enable);
+    /**
+    * Returns weather or not FEC is enabled
+    */
+    bool IsFecEnabled();
+    /**
+    * Returns weather or not NackFec is enabled
+    */
+    bool IsNackFecEnabled();
+    /**
+    * Updates the max pay load size
+    */
+     /**
+    * Enable NackFec and update error resilience parameters
+    */
+    void EnableNackFEC(bool enable);
+
+    void SetMtu(WebRtc_Word32 mtu);
+
+    /*
+    * Get actual input frame rate
+    */
+    WebRtc_UWord32 InputFrameRate();
+
+    /*
+    * Get actual sent frame rate
+    */
+    float SentFrameRate();
+    /*
+    * Get actual sent bit rate
+    */
+    float SentBitRate();
+    /*
+    * Get maximum allowed bit rate
+    */
+    WebRtc_Word32 MaxBitRate();
+    /*
+    * Inform Media Optimization of encoding output: Length and frame type
+    */
+    WebRtc_Word32 UpdateWithEncodedData(WebRtc_Word32 encodedLength,
+                                        FrameType encodedFrameType);
+    /*
+    * Register a protection callback to be used to inform the user about the
+    * protection methods used
+    */
+    WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protectionCallback);
+    /*
+    * Register a quality settings callback to be used to inform VPM/user about the optimal
+    * quality settings (frame rate/dimension) required
+    */
+    WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings);
+    void EnableFrameDropper(bool enable);
+
+    bool DropFrame();
+
+      /*
+    * Get number of key/delta frames encoded
+    */
+    WebRtc_Word32 SentFrameCount(VCMFrameCount &frameCount) const;
+
+    /*
+    *  update incoming frame rate value
+    */
+    void UpdateIncomingFrameRate();
+
+    /**
+    * Update content metric Data
+    */
+    void updateContentData(const VideoContentMetrics* contentMetrics);
+
+    /**
+    * Compute new Quality Mode
+    */
+    WebRtc_Word32 SelectQuality();
+
+private:
+
+    void UpdateBitRateEstimate(WebRtc_Word64 encodedLength, WebRtc_Word64 nowMs);
+    /*
+    * verify if QM settings differ from default, i.e. if an update is required
+    * Compute actual values, as will be sent to the encoder
+    */
+    bool QMUpdate(VCMQualityMode* qm);
+    /**
+    * check if we should make a QM change
+    * will return 1 if yes, 0 otherwise
+    */
+    bool checkStatusForQMchange();
+
+    void ProcessIncomingFrameRate(WebRtc_Word64 now);
+
+    enum { kFrameCountHistorySize = 90};
+    enum { kFrameHistoryWinMs = 2000};
+
+    WebRtc_Word32                     _id;
+
+    WebRtc_Word32                     _maxBitRate;
+    VideoCodecType                    _sendCodecType;
+    WebRtc_UWord16                    _codecWidth;
+    WebRtc_UWord16                    _codecHeight;
+    float                             _userFrameRate;
+
+    VCMFrameDropper*                  _frameDropper;
+    VCMLossProtectionLogic*           _lossProtLogic;
+    WebRtc_UWord32                    _lossProtOverhead;
+    WebRtc_UWord8                     _packetLossEnc;
+    WebRtc_UWord8                     _fractionLost;
+
+
+    WebRtc_UWord32                    _sendStatistics[4];
+    WebRtc_UWord32                    _sendStatisticsZeroEncode;
+    WebRtc_Word32                     _maxPayloadSize;
+    WebRtc_UWord32                    _lastBitRate;
+    WebRtc_UWord32                    _targetBitRate;
+
+    float                             _incomingFrameRate;
+    WebRtc_Word64                     _incomingFrameTimes[kFrameCountHistorySize];
+
+    bool                              _enableQm;
+
+    VCMProtectionCallback*            _videoProtectionCallback;
+    VCMQMSettingsCallback*            _videoQMSettingsCallback;
+
+    VCMEncodedFrameSample             _encodedFrameSamples[kBitrateMaxFrameSamples];
+    float                             _avgSentBitRateBps;
+
+    WebRtc_UWord32                    _keyFrameCnt;
+    WebRtc_UWord32                    _deltaFrameCnt;
+
+    VCMContentMetricsProcessing*      _content;
+    VCMQmSelect*                      _qms;
+
+    WebRtc_Word64                     _lastQMUpdateTime;
+    WebRtc_Word64                     _lastChangeTime; // content or user triggered
+
+
+}; // end of VCMMediaOptimization class definition
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_

modules/video_coding/main/source/nack_fec_tables.h

@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
+
+namespace webrtc
+{
+
+// Table for softening FEC rate for NACK/FEC protection method
+const WebRtc_UWord16 VCMNackFecTable[200] = {
+
+27,
+28,
+30,
+31,
+33,
+35,
+36,
+38,
+40,
+42,
+45,
+47,
+49,
+52,
+54,
+57,
+60,
+63,
+66,
+70,
+73,
+77,
+81,
+85,
+89,
+94,
+98,
+103,
+108,
+114,
+120,
+126,
+132,
+138,
+145,
+152,
+160,
+168,
+176,
+185,
+194,
+203,
+213,
+223,
+234,
+246,
+257,
+270,
+283,
+296,
+310,
+325,
+340,
+356,
+373,
+390,
+408,
+427,
+446,
+467,
+488,
+510,
+532,
+556,
+581,
+606,
+632,
+659,
+688,
+717,
+747,
+778,
+810,
+843,
+877,
+912,
+948,
+985,
+1022,
+1061,
+1101,
+1142,
+1183,
+1226,
+1269,
+1314,
+1359,
+1404,
+1451,
+1498,
+1546,
+1594,
+1643,
+1693,
+1743,
+1793,
+1843,
+1894,
+1945,
+1996,
+2048,
+2099,
+2150,
+2201,
+2252,
+2302,
+2352,
+2402,
+2452,
+2501,
+2549,
+2597,
+2644,
+2691,
+2736,
+2781,
+2826,
+2869,
+2912,
+2953,
+2994,
+3034,
+3073,
+3110,
+3147,
+3183,
+3218,
+3252,
+3285,
+3317,
+3348,
+3378,
+3407,
+3436,
+3463,
+3489,
+3514,
+3539,
+3563,
+3585,
+3607,
+3628,
+3649,
+3668,
+3687,
+3705,
+3722,
+3739,
+3755,
+3770,
+3785,
+3799,
+3812,
+3825,
+3838,
+3849,
+3861,
+3872,
+3882,
+3892,
+3901,
+3910,
+3919,
+3927,
+3935,
+3943,
+3950,
+3957,
+3963,
+3969,
+3975,
+3981,
+3987,
+3992,
+3997,
+4001,
+4006,
+4010,
+4014,
+4018,
+4022,
+4025,
+4029,
+4032,
+4035,
+4038,
+4041,
+4043,
+4046,
+4048,
+4050,
+4053,
+4055,
+4057,
+4059,
+4060,
+4062,
+4064,
+4065,
+4067,
+
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_

modules/video_coding/main/source/packet.cc

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet.h"
+#include "module_common_types.h"
+
+#include <assert.h>
+
+namespace webrtc {
+
+VCMPacket::VCMPacket(const WebRtc_UWord8* ptr,
+                               const WebRtc_UWord32 size,
+                               const WebRtcRTPHeader& rtpHeader) :
+    payloadType(rtpHeader.header.payloadType),
+    timestamp(rtpHeader.header.timestamp),
+    seqNum(rtpHeader.header.sequenceNumber),
+    dataPtr(ptr),
+    sizeBytes(size),
+    markerBit(rtpHeader.header.markerBit),
+
+    frameType(rtpHeader.frameType),
+    codec(kVideoCodecUnknown),
+    isFirstPacket(rtpHeader.type.Video.isFirstPacket),
+    completeNALU(kNaluComplete),
+    insertStartCode(false),
+    bits(false)
+{
+    CopyCodecSpecifics(rtpHeader.type.Video);
+}
+
+VCMPacket::VCMPacket(const WebRtc_UWord8* ptr, WebRtc_UWord32 size, WebRtc_UWord16 seq, WebRtc_UWord32 ts, bool mBit) :
+    payloadType(0),
+    timestamp(ts),
+    seqNum(seq),
+    dataPtr(ptr),
+    sizeBytes(size),
+    markerBit(mBit),
+
+    frameType(kVideoFrameDelta),
+    codec(kVideoCodecUnknown),
+    isFirstPacket(false),
+    completeNALU(kNaluComplete),
+    insertStartCode(false),
+    bits(false)
+{}
+
+void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader)
+{
+    RTPVideoTypeHeader codecHeader = videoHeader.codecHeader;
+    switch(videoHeader.codec)
+    {
+        case kRTPVideoVP8:
+            {
+                codec = kVideoCodecVP8;
+                break;
+            }
+        case kRTPVideoI420:
+            {
+                codec = kVideoCodecI420;
+                break;
+            }
+        default:
+            {
+                codec = kVideoCodecUnknown;
+                break;
+            }
+    }
+}
+
+}

modules/video_coding/main/source/packet.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+#include "jitter_buffer_common.h"
+
+namespace webrtc
+{
+
+class VCMPacket
+{
+public:
+    VCMPacket(const WebRtc_UWord8* ptr,
+                   const WebRtc_UWord32 size,
+                   const WebRtcRTPHeader& rtpHeader);
+    VCMPacket(const WebRtc_UWord8* ptr,
+                   WebRtc_UWord32 size,
+                   WebRtc_UWord16 seqNum,
+                   WebRtc_UWord32 timestamp,
+                   bool markerBit);
+
+    WebRtc_UWord8           payloadType;
+    WebRtc_UWord32          timestamp;
+    WebRtc_UWord16          seqNum;
+    const WebRtc_UWord8*    dataPtr;
+    WebRtc_UWord32          sizeBytes;
+    bool                    markerBit;
+
+    FrameType               frameType;
+    webrtc::VideoCodecType  codec;
+
+    bool isFirstPacket;                 // Is this first packet in a frame.
+    VCMNaluCompleteness completeNALU;   // Default is kNaluIncomplete.
+    bool insertStartCode;               // True if a start code should be inserted before this
+                                        // packet.
+    bool bits;                          // The first bits of this packets are zero and the
+                                        // first
+                                        // byte should be ORed with the last packet of the
+                                        // previous frame.
+
+protected:
+    void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_

modules/video_coding/main/source/qm_select.cc

@@ -0,0 +1,684 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "qm_select.h"
+#include "internal_defines.h"
+#include "qm_select_data.h"
+
+#include "module_common_types.h"
+#include "video_coding_defines.h"
+#include "trace.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+VCMQmSelect::VCMQmSelect()
+{
+    _qm = new VCMQualityMode();
+     Reset();
+}
+
+VCMQmSelect::~VCMQmSelect()
+{
+    delete _qm;
+}
+
+void
+VCMQmSelect::ResetQM()
+{
+    _motion.Reset();
+    _spatial.Reset();
+    _coherence.Reset();
+    _stationaryMotion = 0;
+    _aspectRatio = 1;
+    _maxRateQM = 0;
+    _imageType = 1;
+    _userResolutionPref = 50; // Neutral
+    _qm->Reset();
+    return;
+}
+
+void
+VCMQmSelect::ResetRates()
+{
+    _sumEncodedBytes = 0;
+    _sumTargetRate = 0;
+    _sumIncomingFrameRate = 0;
+    _sumFrameRateMM = 0;
+    _sumSeqRateMM = 0;
+    _frameCnt = 0;
+    _frameCntDelta = 0;
+    _lowBufferCnt = 0;
+    _updateRateCnt = 0;
+    return;
+}
+
+void
+VCMQmSelect::Reset()
+{
+   _stateDecFactorSpatial = 1;
+   _stateDecFactorTemp  = 1;
+   _bufferLevel = 0;
+   _targetBitRate = 0;
+   _incomingFrameRate = 0;
+   _userFrameRate = 0;
+   _perFrameBandwidth =0;
+    ResetQM();
+    ResetRates();
+    return;
+}
+
+//Initialize after reset of encoder
+WebRtc_Word32
+VCMQmSelect::Initialize(float bitRate, float userFrameRate, WebRtc_UWord32 width, WebRtc_UWord32 height)
+{
+    if (userFrameRate == 0.0f || width == 0 || height == 0)
+    {
+        return VCM_PARAMETER_ERROR;
+    }
+    _targetBitRate = bitRate;
+    _userFrameRate = userFrameRate;
+    //Encoder width and height
+    _width = width;
+    _height = height;
+    //Initial buffer level
+    _bufferLevel = INIT_BUFFER_LEVEL * _targetBitRate;
+    if ( _incomingFrameRate == 0 )
+    {
+        _perFrameBandwidth = _targetBitRate / _userFrameRate;
+        _incomingFrameRate = _userFrameRate;
+    }
+    else
+    {
+    //Take average: this is due to delay in update of new frame rate in encoder:
+    //userFrameRate is the new one, incomingFrameRate is the old one (based on previous ~ 1sec)
+        _perFrameBandwidth = 0.5 *( _targetBitRate / _userFrameRate + _targetBitRate / _incomingFrameRate );
+    }
+    _init  = true;
+
+
+    return VCM_OK;
+}
+
+WebRtc_Word32
+VCMQmSelect::SetPreferences(WebRtc_Word8 resolPref)
+{
+    // Preference setting for temporal over spatial resolution
+    // 100 means temporal, 0 means spatial, 50 is neutral (we decide)
+    _userResolutionPref = resolPref;
+
+    return VCM_OK;
+}
+
+//Update after every encoded frame
+void
+VCMQmSelect::UpdateEncodedSize(WebRtc_Word64 encodedSize, FrameType encodedFrameType)
+{
+    //Update encoded size;
+    _sumEncodedBytes += encodedSize;
+    _frameCnt++;
+
+    //Convert to Kbps
+    float encodedSizeKbits = (float)((encodedSize * 8.0) / 1000.0);
+
+    //Update the buffer level: per_frame_BW is updated when encoder is updated, every ~1sec
+    _bufferLevel += _perFrameBandwidth - encodedSizeKbits;
+
+    const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
+                             encodedFrameType != kVideoFrameGolden);
+
+    //Sum the frame mismatch:
+    //Mismatch here is based on difference of actual encoded frame size and per-frame bandwidth, for delta frames
+    //This is a much stronger condition on rate mismatch than sumSeqRateMM
+    // Note: not used in this version
+    /*
+    if (deltaFrame)
+    {
+         _frameCntDelta++;
+         if (encodedSizeKbits > 0)
+            _sumFrameRateMM += (float) (fabs(encodedSizeKbits - _perFrameBandwidth) / encodedSizeKbits);
+    }
+    */
+
+    //Counter for occurrences of low buffer level
+    if (_bufferLevel <= PERC_BUFFER_THR * INIT_BUFFER_LEVEL * _targetBitRate)
+    {
+        _lowBufferCnt++;
+    }
+
+
+
+}
+
+//Update after SetTargetRates in MediaOpt (every ~1sec)
+void
+VCMQmSelect::UpdateRates(float targetBitRate, float avgSentBitRate, float incomingFrameRate)
+{
+
+    //Sum the target bitrate and incoming frame rate: these values are the encoder rates (from previous ~1sec),
+    //i.e, before the update for next ~1sec
+    _sumTargetRate += _targetBitRate;
+    _sumIncomingFrameRate  += _incomingFrameRate;
+    _updateRateCnt++;
+
+    //Convert to kbps
+    float avgSentBitRatekbps = avgSentBitRate / 1000.0f;
+
+    //Sum the sequence rate mismatch:
+    //Mismatch here is based on difference between target rate the encoder used (in previous ~1sec) and the average actual
+    //encoding rate at current time
+    if (fabs(_targetBitRate - avgSentBitRatekbps) <  THRESH_SUM_MM && _targetBitRate > 0.0 )
+        _sumSeqRateMM += (float) (fabs(_targetBitRate - avgSentBitRatekbps) / _targetBitRate );
+
+    //Update QM with the current new target and frame rate: these values are ones the encoder will use for the current/next ~1sec
+    _targetBitRate =  targetBitRate;
+    _incomingFrameRate = incomingFrameRate;
+
+    //Update QM with an (average) encoder per_frame_bandwidth: this is the per_frame_bw for the next ~1sec
+    _perFrameBandwidth  = 0.0f;
+    if (_incomingFrameRate > 0.0f)
+    {
+        _perFrameBandwidth = _targetBitRate / _incomingFrameRate;
+    }
+
+}
+
+WebRtc_Word32
+VCMQmSelect::SelectQuality(const VideoContentMetrics* contentMetrics, VCMQualityMode** qm)
+{
+    if (!_init)
+    {
+        return VCM_UNINITIALIZED;
+    }
+    if (contentMetrics == NULL)
+    {
+        Reset(); //default values
+        *qm =  _qm;
+        return VCM_OK;
+    }
+
+    //Default settings
+    _qm->spatialWidthFact = 1;
+    _qm->spatialHeightFact = 1;
+    _qm->temporalFact = 1;
+
+    _contentMetrics = contentMetrics;
+
+    //Update native values
+    _nativeWidth = _contentMetrics->nativeWidth;
+    _nativeHeight = _contentMetrics->nativeHeight;
+    _nativeFrameRate = _contentMetrics->nativeFrameRate;
+
+    //Aspect ratio: used for selection of 1x2,2x1,2x2
+    _aspectRatio = (float)_width / (float)_height;
+
+    float avgTargetRate = 0.0f;
+    float avgIncomingFrameRate = 0.0f;
+    float ratioBufferLow = 0.0f;
+    float rateMisMatch = 0.0f;
+    if (_frameCnt > 0)
+    {
+        ratioBufferLow = (float)_lowBufferCnt / (float)_frameCnt;
+    }
+    if (_updateRateCnt > 0)
+    {
+        //use seq-rate mismatch for now
+        rateMisMatch = (float)_sumSeqRateMM / (float)_updateRateCnt;
+        //rateMisMatch = (float)_sumFrameRateMM / (float)_frameCntDelta;
+
+        //average target and incoming frame rates
+        avgTargetRate = (float)_sumTargetRate / (float)_updateRateCnt;
+        avgIncomingFrameRate = (float)_sumIncomingFrameRate / (float)_updateRateCnt;
+    }
+
+    //For qm selection below, may want to weight the average encoder rates with the current (for next ~1sec) rate values
+    //uniform average for now:
+    float w1 = 0.5f;
+    float w2 = 0.5f;
+    avgTargetRate = w1 * avgTargetRate + w2 * _targetBitRate;
+    avgIncomingFrameRate = w1 * avgIncomingFrameRate + w2 * _incomingFrameRate;
+
+    //Set the maximum transitional rate and image type: for up-sampled spatial dimensions
+    //Needed to get the transRate for going back up in spatial resolution (only 2x2 allowed in this version)
+    SetMaxRateForQM(2 * _width, 2 * _height);
+    WebRtc_UWord8  imageType2  = _imageType;
+    WebRtc_UWord32 maxRateQM2 = _maxRateQM;
+
+    //Set the maximum transitional rate and image type: for the input/encoder spatial dimensions
+    SetMaxRateForQM(_width, _height);
+
+    //Compute metric features
+    MotionNFD();
+    Spatial();
+
+    //
+    //Get transitional rate from table, based on image type and content class
+    //
+
+    //Get image size class: map _imageType to 2 classes
+    WebRtc_UWord8 imageClass = 1;
+    if (_imageType <= 3) imageClass = 0;
+
+    WebRtc_UWord8 contentClass  = 3 * _motion.level + _spatial.level;
+    WebRtc_UWord8 tableIndex = imageClass * 9 + contentClass;
+    float scaleTransRate = kScaleTransRateQm[tableIndex];
+
+    // for transRate for going back up spatially
+    WebRtc_UWord8 imageClass2 = 1;
+    if (imageType2 <= 3) imageClass2 = 0;
+    WebRtc_UWord8 tableIndex2 = imageClass2 * 9 + contentClass;
+    float scaleTransRate2 = kScaleTransRateQm[tableIndex2];
+    //
+
+    WebRtc_UWord32 estimatedTransRateDown = (WebRtc_UWord32) (_incomingFrameRate * scaleTransRate * _maxRateQM / 30);
+    WebRtc_UWord32 estimatedTransRateUpT =  (WebRtc_UWord32) (TRANS_RATE_SCALE_UP_TEMP * 2 * _incomingFrameRate * scaleTransRate * _maxRateQM / 30);
+    WebRtc_UWord32 estimatedTransRateUpS =  (WebRtc_UWord32) (TRANS_RATE_SCALE_UP_SPATIAL * _incomingFrameRate * scaleTransRate2 * maxRateQM2 / 30);
+
+    //
+    //done with transitional rate
+    //
+
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, -1,
+                   "Content Metrics: Motion = %d , Spatial = %d, Est. Trans. BR = %d",
+                   _motion.level, _spatial.level, estimatedTransRateDown);
+
+
+
+    //
+    //CHECK FOR GOING BACK UP IN RESOLUTION
+    //
+    bool selectedUp = false;
+    //Check if native has been spatially down-sampled
+    if (_stateDecFactorSpatial > 1)
+    {
+        //check conditions on frame_skip and rate_mismatch
+        if ( (avgTargetRate > estimatedTransRateUpS) &&
+             (ratioBufferLow < MAX_BUFFER_LOW) && (rateMisMatch < MAX_RATE_MM) )
+        {
+            //width/height scaled back up: setting 0 indicates scaling back to native
+            _qm->spatialHeightFact = 0;
+            _qm->spatialWidthFact = 0;
+            selectedUp = true;
+        }
+    }
+    //Check if native has been temporally down-sampled
+    if (_stateDecFactorTemp > 1)
+    {
+        if ( (avgTargetRate > estimatedTransRateUpT) &&
+             (ratioBufferLow < MAX_BUFFER_LOW) && (rateMisMatch < MAX_RATE_MM) )
+        {
+            //temporal scale back up: setting 0 indicates scaling back to native
+            _qm->temporalFact = 0;
+            selectedUp = true;
+        }
+    }
+
+    //leave QM if we selected to go back up in either spatial or temporal resolution
+    if (selectedUp == true)
+    {
+        //Update down-sampling state
+        //Note: only temp reduction by 2 is allowed
+        if (_qm->temporalFact == 0)
+        {
+            _stateDecFactorTemp = _stateDecFactorTemp / 2;
+        }
+        //Update down-sampling state
+        //Note: only spatial reduction by 2x2 is allowed
+        if (_qm->spatialHeightFact == 0 && _qm->spatialWidthFact == 0 )
+        {
+            _stateDecFactorSpatial = _stateDecFactorSpatial / 4;
+        }
+       *qm = _qm;
+       return VCM_OK;
+    }
+
+    //
+    //done with checking for going back up
+    //
+
+    //
+    //CHECK FOR RESOLUTION REDUCTION
+    //
+
+    //ST QM extraction if:
+    // (1) target rate is lower than transitional rate (with safety margin), or
+    // (2) frame skip is larger than threshold, or
+    // (3) rate mismatch is larger than threshold
+
+    if ( (avgTargetRate < estimatedTransRateDown ) || (ratioBufferLow > MAX_BUFFER_LOW)
+         || (rateMisMatch > MAX_RATE_MM) )
+    {
+
+        WebRtc_UWord8 spatialFact = 1;
+        WebRtc_UWord8 tempFact = 1;
+
+        //Get the Action:
+        //Note: only consider spatial by 2x2 OR temporal reduction by 2 in this version
+        if (_motion.level == kLow && _spatial.level == kLow)
+        {
+            spatialFact = 1;
+            tempFact = 1;
+        }
+        else if (_motion.level == kLow && _spatial.level == kHigh)
+        {
+            spatialFact = 1;
+            tempFact = 2;
+        }
+        else if (_motion.level == kLow && _spatial.level == kDefault)
+        {
+            spatialFact = 1;
+            tempFact = 2;
+        }
+        else if (_motion.level == kHigh && _spatial.level == kLow)
+        {
+            spatialFact = 4;
+            tempFact = 1;
+        }
+        else if (_motion.level == kHigh && _spatial.level == kHigh)
+        {
+            spatialFact = 1;
+            tempFact = 2;
+        }
+        else if (_motion.level == kHigh && _spatial.level == kDefault)
+        {
+            spatialFact = 4;
+            tempFact = 1;
+        }
+        else if (_motion.level == kDefault && _spatial.level == kLow)
+        {
+            spatialFact = 4;
+            tempFact = 1;
+        }
+        else if (_motion.level == kDefault && _spatial.level == kHigh)
+        {
+            spatialFact = 1;
+            tempFact = 2;
+        }
+        else if (_motion.level == kDefault && _spatial.level == kDefault)
+        {
+            spatialFact = 1;
+            tempFact = 1;
+        }
+        //
+        switch(spatialFact)
+        {
+        case 4:
+            _qm->spatialWidthFact = 2;
+            _qm->spatialHeightFact = 2;
+            break;
+        case 2:
+             //default is 1x2 (H)
+            _qm->spatialWidthFact = 2;
+            _qm->spatialHeightFact = 1;
+            //Select 1x2,2x1, or back to 2x2: depends on prediction errors, aspect ratio, and horizontalness of motion
+            //Note: directional selection not used in this version
+            //SelectSpatialDirectionMode((float) estimatedTransRateDown);
+            break;
+        default:
+            _qm->spatialWidthFact = 1;
+            _qm->spatialHeightFact = 1;
+            break;
+        }
+        _qm->temporalFact = tempFact;
+
+        //Sanity check on ST QM selection: override the settings for too small image size and frame rate
+        //Also check limit the current down-sampling state
+
+        //No spatial sampling if image size is too small (QCIF)
+        if ( (_width * _height) <= MIN_IMAGE_SIZE  || _stateDecFactorSpatial >= MAX_SPATIAL_DOWN_FACT)
+        {
+            _qm->spatialWidthFact = 1;
+            _qm->spatialHeightFact = 1;
+        }
+
+        //No frame rate reduction below some point: use the (average) incoming frame rate
+        if ( avgIncomingFrameRate <= MIN_FRAME_RATE_QM  || _stateDecFactorTemp >= MAX_TEMP_DOWN_FACT)
+        {
+            _qm->temporalFact = 1;
+        }
+
+        //No down-sampling if current spatial-temporal downsampling state is above threshold
+        if (_stateDecFactorTemp * _stateDecFactorSpatial >= MAX_SPATIAL_TEMP_DOWN_FACT)
+        {
+            _qm->spatialWidthFact = 1;
+            _qm->spatialHeightFact = 1;
+            _qm->temporalFact = 1;
+        }
+        //
+        //done with sanity checks on ST QM selection
+        //
+
+        //Note: to disable spatial down-sampling
+        // _qm->spatialWidthFact = 1;
+        // _qm->spatialHeightFact = 1;
+
+        //Update down-sampling states
+        _stateDecFactorSpatial = _stateDecFactorSpatial * _qm->spatialWidthFact * _qm->spatialHeightFact;
+        _stateDecFactorTemp = _stateDecFactorTemp * _qm->temporalFact;
+
+    }
+    else
+    {
+      *qm = _qm;
+      return VCM_OK;
+    }
+    // done with checking for resolution reduction
+
+    *qm = _qm;
+    return VCM_OK;
+
+
+}
+
+WebRtc_Word32
+VCMQmSelect::SelectSpatialDirectionMode(float transRate)
+{
+    //Default is 1x2 (H)
+
+    //For bit rates well below transitional rate, we select 2x2
+    if ( _targetBitRate < transRate * RATE_RED_SPATIAL_2X2 )
+    {
+        _qm->spatialWidthFact = 2;
+        _qm->spatialHeightFact = 2;
+        return VCM_OK;
+    }
+
+    //Otherwise check prediction errors, aspect ratio, horizonalness of motion
+
+    float spatialErr = _contentMetrics->spatialPredErr;
+    float spatialErrH = _contentMetrics->spatialPredErrH;
+    float spatialErrV = _contentMetrics->spatialPredErrV;
+
+    //favor 1x2 if aspect_ratio is 16:9
+    if (_aspectRatio >= 16.0f / 9.0f )
+    {
+        //check if 1x2 has lowest prediction error
+        if (spatialErrH < spatialErr && spatialErrH < spatialErrV)
+        {
+            return VCM_OK;
+        }
+    }
+
+    //check for 2x2 selection: favor 2x2 over 1x2 and 2x1
+    if (spatialErr < spatialErrH * (1.0f + SPATIAL_ERR_2X2_VS_H) &&
+        spatialErr < spatialErrV * (1.0f + SPATIAL_ERR_2X2_VS_V))
+    {
+        _qm->spatialWidthFact = 2;
+        _qm->spatialHeightFact = 2;
+         return VCM_OK;
+    }
+
+    //check for 2x1 selection:
+    if (spatialErrV < spatialErrH * (1.0f - SPATIAL_ERR_V_VS_H) &&
+        spatialErrV < spatialErr * (1.0f - SPATIAL_ERR_2X2_VS_V))
+    {
+        _qm->spatialWidthFact = 1;
+        _qm->spatialHeightFact = 2;
+         return VCM_OK;
+    }
+
+    return VCM_OK;
+}
+
+void
+VCMQmSelect::Coherence()
+{
+    float horizNZ  = _contentMetrics->motionHorizontalness;
+    float distortionNZ  = _contentMetrics->motionClusterDistortion;
+
+    //Coherence measure: combine horizontalness with cluster distortion
+    _coherence.value = COH_MAX;
+    if (distortionNZ > 0.)
+    {
+        _coherence.value = horizNZ / distortionNZ;
+    }
+    _coherence.value = VCM_MIN(COH_MAX, _coherence.value);
+
+    if (_coherence.value < COHERENCE_THR)
+    {
+        _coherence.level = kLow;
+    }
+    else
+    {
+        _coherence.level = kHigh;
+    }
+
+}
+
+void
+VCMQmSelect::MotionNFD()
+{
+    _motion.value = _contentMetrics->motionMagnitudeNZ;
+
+    // determine motion level
+    if (_motion.value < LOW_MOTION_NFD)
+    {
+        _motion.level = kLow;
+    }
+    else if (_motion.value > HIGH_MOTION_NFD)
+    {
+        _motion.level  = kHigh;
+    }
+    else
+    {
+        _motion.level = kDefault;
+    }
+
+}
+
+void
+VCMQmSelect::Motion()
+{
+
+    float sizeZeroMotion = _contentMetrics->sizeZeroMotion;
+    float motionMagNZ = _contentMetrics->motionMagnitudeNZ;
+
+    //take product of size and magnitude with equal weight for now
+    _motion.value = (1.0f - sizeZeroMotion) * motionMagNZ;
+
+    //stabilize: motionMagNZ could be large when only few motion blocks are non-zero
+    _stationaryMotion = false;
+    if (sizeZeroMotion > HIGH_ZERO_MOTION_SIZE)
+    {
+        _motion.value = 0.0f;
+        _stationaryMotion = true;
+    }
+    // determine motion level
+    if (_motion.value < LOW_MOTION)
+    {
+        _motion.level = kLow;
+    }
+    else if (_motion.value > HIGH_MOTION)
+    {
+        _motion.level  = kHigh;
+    }
+    else
+    {
+        _motion.level = kDefault;
+    }
+}
+
+
+void
+VCMQmSelect::Spatial()
+{
+    float spatialErr =  _contentMetrics->spatialPredErr;
+    float spatialErrH = _contentMetrics->spatialPredErrH;
+    float spatialErrV = _contentMetrics->spatialPredErrV;
+    //Spatial measure: take average of 3 prediction errors
+    _spatial.value = (spatialErr + spatialErrH + spatialErrV) / 3.0f;
+
+    float scale = 1.0f;
+    //Reduce thresholds for HD scenes
+    if (_imageType > 3)
+    {
+        scale = (float)SCALE_TEXTURE_HD;
+    }
+
+    if (_spatial.value > scale * HIGH_TEXTURE)
+    {
+        _spatial.level = kHigh;
+    }
+    else if (_spatial.value < scale * LOW_TEXTURE)
+    {
+        _spatial.level = kLow;
+    }
+    else
+    {
+         _spatial.level = kDefault;
+    }
+
+
+}
+
+
+WebRtc_Word32
+VCMQmSelect::SetMaxRateForQM(WebRtc_UWord32 width, WebRtc_UWord32 height)
+{
+    // Match image type
+    WebRtc_UWord32 imageSize = width * height;
+
+    if (imageSize < kFrameSizeTh[0])
+    {
+        _imageType  = 0;
+    }
+    else if (imageSize < kFrameSizeTh[1])
+    {
+        _imageType  = 1;
+    }
+    else if (imageSize < kFrameSizeTh[2])
+    {
+        _imageType  = 2;
+    }
+    else if (imageSize < kFrameSizeTh[3])
+    {
+        _imageType  = 3;
+    }
+    else if (imageSize < kFrameSizeTh[4])
+    {
+        _imageType  = 4;
+    }
+    else if (imageSize < kFrameSizeTh[5])
+    {
+        _imageType  = 5;
+    }
+    else
+    {
+        _imageType  = 6;
+    }
+
+    // set max rate based on image size
+    _maxRateQM = kMaxRateQm[_imageType];
+
+    return VCM_OK;
+}
+
+}

modules/video_coding/main/source/qm_select.h

@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
+#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
+
+#include "typedefs.h"
+#include "common_types.h"
+/************************/
+/* Quality Modes       */
+/**********************/
+
+namespace webrtc
+{
+
+struct VideoContentMetrics;
+
+struct VCMQualityMode
+{
+    VCMQualityMode():spatialWidthFact(1), spatialHeightFact(1), temporalFact(1){}
+    void Reset()
+    {
+        spatialWidthFact = 1;
+        spatialHeightFact = 1;
+        temporalFact = 1;
+    }
+
+    WebRtc_UWord16  spatialWidthFact;
+    WebRtc_UWord16  spatialHeightFact;
+    WebRtc_UWord16  temporalFact;
+};
+
+enum VCMMagValues
+{
+    kLow,
+    kHigh,
+    kDefault //default do nothing mode
+};
+
+struct VCMContFeature
+{
+    VCMContFeature(): value(0.0f), level(kDefault){}
+
+    void Reset()
+    {
+        value = 0.0f;
+        level = kDefault;
+    }
+
+    float value;
+    VCMMagValues level;
+};
+
+class VCMQmSelect
+{
+public:
+    VCMQmSelect();
+    ~VCMQmSelect();
+
+    // Initialize:
+    WebRtc_Word32 Initialize(float bitRate, float userFrameRate, WebRtc_UWord32 width, WebRtc_UWord32 height);
+
+    // Allow the user to set preferences: favor frame rate/resolution
+    WebRtc_Word32 SetPreferences(WebRtc_Word8 resolPref);
+
+    // Extract ST QM behavior and make decision
+    // Inputs: Content Metrics per frame (averaged over time)
+    //         qm: Reference to the quality modes pointer
+    WebRtc_Word32 SelectQuality(const VideoContentMetrics* contentMetrics, VCMQualityMode** qm);
+
+    // Update QMselect with actual bit rate (size of the latest encoded frame) and frame type
+    // -> update buffer level and frame-mismatch
+    void UpdateEncodedSize(WebRtc_Word64 encodedSize, FrameType encodedFrameType);
+
+    // Update QM with new rates from SetTargetRates
+    void UpdateRates(float targetBitRate, float avgSentRate, float incomingFrameRate);
+
+    // Select 1x2,2x2,2x2 spatial sampling mode
+    WebRtc_Word32 SelectSpatialDirectionMode(float transRate);
+
+    // Reset values prior to QMSelect
+    void ResetQM();
+
+    // Reset rate quantities and counter values after every QMSelect call
+    void ResetRates();
+
+    // Reset all
+    void Reset();
+private:
+
+    // Compute spatial texture magnitude and level
+    void Spatial();
+
+    // Compute motion magnitude and level
+    void Motion();
+
+    // Compute motion magnitude and level for NFD metric
+    void MotionNFD();
+
+    // Compute coherence magnitude and level
+    void Coherence();
+
+    // Set the max rate for QM selection
+    WebRtc_Word32 SetMaxRateForQM(WebRtc_UWord32 width, WebRtc_UWord32 height);
+
+    // Content Data
+    const VideoContentMetrics*    _contentMetrics;
+
+    // Encoder stats/rate-control metrics
+    float                        _targetBitRate;
+    float                        _userFrameRate;
+    float                        _incomingFrameRate;
+    float                        _perFrameBandwidth;
+    float                        _bufferLevel;
+    float                        _sumTargetRate;
+    float                        _sumIncomingFrameRate;
+    float                        _sumSeqRateMM;
+    float                        _sumFrameRateMM;
+    WebRtc_Word64                _sumEncodedBytes;
+
+    //Encoder and native frame sizes
+    WebRtc_UWord32               _width;
+    WebRtc_UWord32               _height;
+    WebRtc_UWord32               _nativeWidth;
+    WebRtc_UWord32               _nativeHeight;
+    WebRtc_UWord8                _stateDecFactorSpatial;
+
+    WebRtc_UWord32               _nativeFrameRate;
+    WebRtc_UWord8                _stateDecFactorTemp;
+
+    //Counters
+    WebRtc_UWord32               _frameCnt;
+    WebRtc_UWord32               _frameCntDelta;
+    WebRtc_UWord32               _updateRateCnt;
+    WebRtc_UWord32               _lowBufferCnt;
+
+    //Content L/M/H values
+    VCMContFeature               _motion;
+    VCMContFeature               _spatial;
+    VCMContFeature               _coherence;
+    bool                         _stationaryMotion;
+
+    //aspect ratio
+    float                        _aspectRatio;
+
+    //Max rate to saturate the transitionalRate
+    WebRtc_UWord32               _maxRateQM;
+    WebRtc_UWord8                _imageType;
+
+    //User preference for resolution or qmax change
+    WebRtc_UWord8                _userResolutionPref;
+    bool                         _init;
+    VCMQualityMode*              _qm;
+
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_

modules/video_coding/main/source/qm_select_data.h

@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+
+/***************************************************************
+*QMSelectData.h
+* This file includes parameters used by the Quality Modes selection process
+****************************************************************/
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+//Initial level of buffer in secs: should corresponds to wrapper settings
+#define INIT_BUFFER_LEVEL 0.5
+
+//
+//PARAMETERS FOR QM SELECTION
+//
+
+//Threshold of (max) buffer size below which we consider too low (underflow)
+#define PERC_BUFFER_THR  0.10
+
+//Threshold on rate mismatch
+#define MAX_RATE_MM  0.5
+
+//Threshold on the occurrences of low buffer levels
+#define MAX_BUFFER_LOW 0.5
+
+//Factor for transitional rate for going back up in resolution
+#define TRANS_RATE_SCALE_UP_SPATIAL     1.25
+#define TRANS_RATE_SCALE_UP_TEMP        1.25
+
+//Maximum possible transitional rate: (units in kbps), for 30fps
+const WebRtc_UWord16 kMaxRateQm[7] = {
+    100,  //QCIF
+    500, //CIF
+    800, //VGA
+    1500, //4CIF
+    2000, //720 HD 4:3,
+    2500, //720 HD 16:9
+    3000  //1080HD
+};
+
+//Scale for transitional rate: based on content class
+// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
+const float kScaleTransRateQm[18] = {
+    //4CIF and lower
+    0.25f,       // L, L
+    0.75f,       // L, H
+    0.75f,       // L, D
+    0.75f,       // H ,L
+    0.50f,       // H, H
+    0.50f,       // H, D
+    0.50f,       // D, L
+    0.625f,       // D, D
+    0.25f,       // D, H
+
+    //over 4CIF: WHD, HD
+    0.25f,       // L, L
+    0.75f,       // L, H
+    0.75f,       // L, D
+    0.75f,       // H ,L
+    0.50f,       // H, H
+    0.50f,       // H, D
+    0.50f,       // D, L
+    0.625f,       // D, D
+    0.25f        // D, H
+};
+
+//Control the total amount of down-sampling allowed
+#define MAX_SPATIAL_DOWN_FACT       4
+#define MAX_TEMP_DOWN_FACT          4
+#define MAX_SPATIAL_TEMP_DOWN_FACT  8
+
+//
+//
+//
+
+//PARAMETETS FOR SETTING LOW/HIGH VALUES OF METRICS:
+//
+//Threshold to determine if high amount of zero_motion
+#define HIGH_ZERO_MOTION_SIZE 0.95
+
+//Thresholds for motion: motion level is derived from motion vectors: motion = size_nz*magn_nz
+#define HIGH_MOTION 0.7
+#define LOW_MOTION  0.4
+
+//Thresholds for motion: motion level is from NFD
+#define HIGH_MOTION_NFD 0.075
+#define LOW_MOTION_NFD  0.04
+
+//Thresholds for spatial prediction error: this is appLied on the min(2x2,1x2,2x1)
+#define HIGH_TEXTURE 0.035
+#define LOW_TEXTURE  0.025
+
+//Used to reduce thresholds for HD scenes: correction factor since higher
+//correlation in HD scenes means lower spatial prediction error
+#define SCALE_TEXTURE_HD 0.9;
+
+//Thresholds for distortion and horizontalness: applied on product: horiz_nz/dist_nz
+#define COHERENCE_THR   1.0
+#define COH_MAX 10
+//
+//
+#define RATE_RED_SPATIAL_2X2    0.6  //percentage reduction in transitional bitrate where 2x2 is selected over 1x2/2x1
+#define SPATIAL_ERR_2X2_VS_H    0.1  //percentage to favor 2x2
+#define SPATIAL_ERR_2X2_VS_V    0.1  //percentage to favor 2x2 over V
+#define SPATIAL_ERR_V_VS_H      0.1  //percentage to favor H over V
+
+
+//Minimum image size for a spatial mode selection: no spatial down-sampling if input size <= MIN_IMAGE_SIZE
+#define MIN_IMAGE_SIZE  25344 //176*144
+
+//Minimum frame rate for temporal mode: no frame rate reduction if incomingFrameRate <= MIN_FRAME_RATE
+#define MIN_FRAME_RATE_QM  8
+
+//Avoid outliers in seq-rate MM
+#define THRESH_SUM_MM 1000
+
+const WebRtc_UWord32 kFrameSizeTh[6] = {
+    // boundaries for the closest standard frame size
+    63360,    //between 176*144 and 352*288
+    204288,   //between 352*288 and 640*480
+    356352,   //between 640*480 and 704*576
+    548352,   //between 704*576 and 960*720
+    806400,   //between 960*720 and 1280*720
+    1497600,  // between 1280*720 and 1920*1080
+};
+
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_

modules/video_coding/main/source/receiver.cc

@@ -0,0 +1,472 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "video_coding.h"
+#include "trace.h"
+#include "encoded_frame.h"
+#include "internal_defines.h"
+#include "receiver.h"
+#include "tick_time.h"
+
+#include <assert.h>
+
+namespace webrtc {
+
+VCMReceiver::VCMReceiver(VCMTiming& timing,
+                         WebRtc_Word32 vcmId,
+                         WebRtc_Word32 receiverId,
+                         bool master)
+:
+_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_vcmId(vcmId),
+_receiverId(receiverId),
+_master(master),
+_jitterBuffer(vcmId, receiverId, master),
+_timing(timing),
+_renderWaitEvent(*new VCMEvent()),
+_nackMode(kNoNack),
+_state(kPassive)
+{
+}
+
+VCMReceiver::~VCMReceiver()
+{
+    _renderWaitEvent.Set();
+    delete &_renderWaitEvent;
+    delete &_critSect;
+}
+
+WebRtc_Word32
+VCMReceiver::Initialize()
+{
+    CriticalSectionScoped cs(_critSect);
+    if (!_jitterBuffer.Running())
+    {
+        _jitterBuffer.Start();
+    }
+    else
+    {
+        _jitterBuffer.Flush();
+    }
+    _renderWaitEvent.Reset();
+    if (_master)
+    {
+        _state = kReceiving;
+    }
+    else
+    {
+        _state = kPassive;
+        SetNackMode(kNoNack);
+    }
+    return VCM_OK;
+}
+
+void VCMReceiver::UpdateRtt(WebRtc_UWord32 rtt)
+{
+    _jitterBuffer.UpdateRtt(rtt);
+}
+
+WebRtc_Word32
+VCMReceiver::InsertPacket(const VCMPacket& packet,
+                          WebRtc_UWord16 frameWidth,
+                          WebRtc_UWord16 frameHeight)
+{
+    // Find an empty frame
+    VCMEncodedFrame *buffer = NULL;
+    const WebRtc_Word32 error = _jitterBuffer.GetFrame(packet, buffer);
+    if (error == VCM_OLD_PACKET_ERROR)
+    {
+        return VCM_OK;
+    }
+    else if (error < 0)
+    {
+        return error;
+    }
+
+    {
+        CriticalSectionScoped cs(_critSect);
+
+        if (frameWidth && frameHeight)
+        {
+            buffer->SetEncodedSize(static_cast<WebRtc_UWord32>(frameWidth),
+                                   static_cast<WebRtc_UWord32>(frameHeight));
+        }
+
+        if (_master)
+        {
+            // Only trace the primary receiver to make it possible
+            // to parse and plot the trace file.
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                       "Packet seqNo %u of frame %u at %u",
+                       packet.seqNum, packet.timestamp,
+                       MaskWord64ToUWord32(VCMTickTime::MillisecondTimestamp()));
+        }
+
+        const bool emptyFrame = (buffer->Length() == 0);
+        const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
+
+        WebRtc_Word64 renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
+
+        if(renderTimeMs < 0)
+        {
+            // Render time error. Assume that this is due to some change in
+            // the incoming video stream and reset the JB and the timing.
+            _jitterBuffer.Flush();
+            _timing.Reset();
+            return VCM_OK;
+        }
+        else if (renderTimeMs < nowMs - kMaxVideoDelayMs)
+        {
+            WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                "This frame should have been rendered more than %u ms ago."
+                "Flushing jitter buffer and resetting timing.", kMaxVideoDelayMs);
+            _jitterBuffer.Flush();
+            _timing.Reset();
+            return VCM_OK;
+        }
+        else if (_timing.TargetVideoDelay() > kMaxVideoDelayMs)
+        {
+            WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                "More than %u ms target delay. Flushing jitter buffer and resetting timing.",
+                kMaxVideoDelayMs);
+            _jitterBuffer.Flush();
+            _timing.Reset();
+            return VCM_OK;
+        }
+
+        // First packet received belonging to this frame.
+        if (buffer->Length() == 0)
+        {
+            const WebRtc_Word64 nowMs = VCMTickTime::MillisecondTimestamp();
+            if (_master)
+            {
+                // Only trace the primary receiver to make it possible to parse and plot the trace file.
+                WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                           "First packet of frame %u at %u", packet.timestamp,
+                           MaskWord64ToUWord32(nowMs));
+            }
+            renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
+            if (renderTimeMs >= 0)
+            {
+                buffer->SetRenderTime(renderTimeMs);
+            }
+            else
+            {
+                buffer->SetRenderTime(nowMs);
+            }
+        }
+
+        // Insert packet into jitter buffer
+        const VCMFrameBufferEnum ret = _jitterBuffer.InsertPacket(buffer, packet);
+
+        if (ret < 0)
+        {
+            WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                       "Error inserting packet seqNo=%u, timeStamp=%u",
+                       packet.seqNum, packet.timestamp);
+            return VCM_JITTER_BUFFER_ERROR;
+        }
+    }
+    return VCM_OK;
+}
+
+VCMEncodedFrame*
+VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs, WebRtc_Word64& nextRenderTimeMs, bool renderTiming, VCMReceiver* dualReceiver)
+{
+    // No need to enter the critical section here since the jitter buffer
+    // is thread-safe.
+    FrameType incomingFrameType = kVideoFrameDelta;
+    nextRenderTimeMs = -1;
+    const WebRtc_Word64 startTimeMs = VCMTickTime::MillisecondTimestamp();
+    WebRtc_Word64 ret = _jitterBuffer.GetNextTimeStamp(maxWaitTimeMs,
+                                                       incomingFrameType,
+                                                       nextRenderTimeMs);
+    if (ret < 0)
+    {
+        // No timestamp in jitter buffer at the moment
+        return NULL;
+    }
+    const WebRtc_UWord32 timeStamp = static_cast<WebRtc_UWord32>(ret);
+
+    // Update the timing
+    _timing.SetRequiredDelay(_jitterBuffer.GetEstimatedJitterMS());
+    _timing.UpdateCurrentDelay(timeStamp);
+
+    const WebRtc_Word32 tempWaitTime = maxWaitTimeMs -
+            static_cast<WebRtc_Word32>(VCMTickTime::MillisecondTimestamp() - startTimeMs);
+    WebRtc_UWord16 newMaxWaitTime = static_cast<WebRtc_UWord16>(VCM_MAX(tempWaitTime, 0));
+
+    VCMEncodedFrame* frame = NULL;
+
+    if (renderTiming)
+    {
+        frame = FrameForDecoding(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
+    }
+    else
+    {
+        frame = FrameForRendering(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
+    }
+
+    if (frame != NULL)
+    {
+        bool retransmitted = false;
+        const WebRtc_Word64 lastPacketTimeMs =
+                _jitterBuffer.LastPacketTime(frame, retransmitted);
+        if (lastPacketTimeMs >= 0 && !retransmitted)
+        {
+            // We don't want to include timestamps which have suffered from retransmission
+            // here, since we compensate with extra retransmission delay within
+            // the jitter estimate.
+            _timing.IncomingTimestamp(timeStamp, lastPacketTimeMs);
+        }
+        if (dualReceiver != NULL)
+        {
+            dualReceiver->UpdateState(*frame);
+        }
+    }
+    return frame;
+}
+
+VCMEncodedFrame*
+VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+                              WebRtc_Word64 nextRenderTimeMs,
+                              VCMReceiver* dualReceiver)
+{
+    // How long can we wait until we must decode the next frame
+    WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
+                                                       VCMTickTime::MillisecondTimestamp());
+
+    // Try to get a complete frame from the jitter buffer
+    VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
+
+    if (frame == NULL && maxWaitTimeMs == 0 && waitTimeMs > 0)
+    {
+        // If we're not allowed to wait for frames to get complete we must calculate if
+        // it's time to decode, and if it's not we will just return for now.
+        return NULL;
+    }
+
+    if (frame == NULL)
+    {
+        // Wait for a complete frame
+        waitTimeMs = VCM_MIN(waitTimeMs, maxWaitTimeMs);
+        frame = _jitterBuffer.GetCompleteFrameForDecoding(waitTimeMs);
+    }
+    if (frame == NULL)
+    {
+        // Get an incomplete frame
+        if (_timing.MaxWaitingTime(nextRenderTimeMs, VCMTickTime::MillisecondTimestamp()) > 0)
+        {
+            // Still time to wait for a complete frame
+            return NULL;
+        }
+
+        // No time left to wait, we must decode this frame now.
+        const bool dualReceiverEnabledAndPassive = dualReceiver != NULL &&
+                                                    dualReceiver->State() == kPassive &&
+                                                    dualReceiver->NackMode() == kNackInfinite;
+        if (dualReceiverEnabledAndPassive && !_jitterBuffer.CompleteSequenceWithNextFrame())
+        {
+            // Jitter buffer state might get corrupt with this frame.
+            dualReceiver->CopyJitterBufferStateFromReceiver(*this);
+        }
+
+        frame = _jitterBuffer.GetFrameForDecoding();
+    }
+    return frame;
+}
+
+VCMEncodedFrame*
+VCMReceiver::FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
+                               WebRtc_Word64 nextRenderTimeMs,
+                               VCMReceiver* dualReceiver)
+{
+    // How long MUST we wait until we must decode the next frame. This is different for the case
+    // where we have a renderer which can render at a specified time. Here we must wait as long
+    // as possible before giving the frame to the decoder, which will render the frame as soon
+    // as it has been decoded.
+    WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
+                                                       VCMTickTime::MillisecondTimestamp());
+    if (maxWaitTimeMs < waitTimeMs)
+    {
+        // If we're not allowed to wait until the frame is supposed to be rendered
+        // we will have to return NULL for now.
+        return NULL;
+    }
+    // Wait until it's time to render
+    _renderWaitEvent.Wait(waitTimeMs);
+
+    // Get a complete frame if possible
+    VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
+
+    if (frame == NULL)
+    {
+        // Get an incomplete frame
+        const bool dualReceiverEnabledAndPassive = dualReceiver != NULL &&
+                                                   dualReceiver->State() == kPassive &&
+                                                   dualReceiver->NackMode() == kNackInfinite;
+        if (dualReceiverEnabledAndPassive && !_jitterBuffer.CompleteSequenceWithNextFrame())
+        {
+            // Jitter buffer state might get corrupt with this frame.
+            dualReceiver->CopyJitterBufferStateFromReceiver(*this);
+        }
+
+        frame = _jitterBuffer.GetFrameForDecoding();
+    }
+    return frame;
+}
+
+void
+VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame)
+{
+    _jitterBuffer.ReleaseFrame(frame);
+}
+
+WebRtc_Word32
+VCMReceiver::ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate)
+{
+    const WebRtc_Word32 ret = _jitterBuffer.GetUpdate(frameRate, bitRate);
+    bitRate /= 1000; // Should be in kbps
+    return ret;
+}
+
+WebRtc_Word32
+VCMReceiver::ReceivedFrameCount(VCMFrameCount& frameCount) const
+{
+    return _jitterBuffer.GetFrameStatistics(frameCount.numDeltaFrames,
+                                            frameCount.numKeyFrames);
+}
+
+void
+VCMReceiver::SetNackMode(VCMNackMode nackMode)
+{
+    CriticalSectionScoped cs(_critSect);
+    _nackMode = nackMode;
+    switch (_nackMode)
+    {
+    case kNackInfinite:
+        {
+            _jitterBuffer.SetNackStatus(true);
+            break;
+        }
+    case kNoNack:
+        {
+            _jitterBuffer.SetNackStatus(false);
+            break;
+        }
+    }
+    if (!_master)
+    {
+        _state = kPassive; // The dual decoder defaults to passive
+    }
+}
+
+VCMNackMode
+VCMReceiver::NackMode() const
+{
+    CriticalSectionScoped cs(_critSect);
+    return _nackMode;
+}
+
+VCMNackStatus
+VCMReceiver::NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size)
+{
+    bool extended = false;
+    WebRtc_UWord16 nackListSize = 0;
+    WebRtc_UWord16* internalNackList = _jitterBuffer.GetNackList(nackListSize, extended);
+    if (internalNackList == NULL && nackListSize == 0xffff)
+    {
+        // This combination is used to trigger key frame requests.
+        size = 0;
+        return kNackKeyFrameRequest;
+    }
+    if (nackListSize > size)
+    {
+        size = nackListSize;
+        return kNackNeedMoreMemory;
+    }
+    memcpy(nackList, internalNackList, nackListSize * sizeof(WebRtc_UWord16));
+    size = nackListSize;
+    return kNackOk;
+}
+
+// Decide whether we should change decoder state. This should be done if the dual decoder
+// has caught up with the decoder decoding with packet losses.
+bool
+VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const
+{
+    if (dualFrame == NULL)
+    {
+        return false;
+    }
+    if (_jitterBuffer.LastDecodedTimestamp() == dualFrame->TimeStamp())
+    {
+        dualReceiver.UpdateState(kWaitForPrimaryDecode);
+        return true;
+    }
+    return false;
+}
+
+void
+VCMReceiver::CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver)
+{
+    _jitterBuffer = receiver._jitterBuffer;
+
+    {
+        CriticalSectionScoped cs(_critSect);
+        if (_nackMode != kNoNack)
+        {
+            _jitterBuffer.SetNackStatus(true);
+        }
+    }
+}
+
+VCMReceiverState
+VCMReceiver::State() const
+{
+    CriticalSectionScoped cs(_critSect);
+    return _state;
+}
+
+void
+VCMReceiver::UpdateState(VCMReceiverState newState)
+{
+    CriticalSectionScoped cs(_critSect);
+    assert(!(_state == kPassive && newState == kWaitForPrimaryDecode));
+//    assert(!(_state == kReceiving && newState == kPassive));
+    _state = newState;
+}
+
+void
+VCMReceiver::UpdateState(VCMEncodedFrame& frame)
+{
+    if (_nackMode == kNoNack)
+    {
+        // Dual decoder mode has not been enabled.
+        return;
+    }
+    // Update the dual receiver state
+    if (frame.Complete() && frame.FrameType() == kVideoFrameKey)
+    {
+        UpdateState(kPassive);
+    }
+    if (State() == kWaitForPrimaryDecode &&
+        frame.Complete() && !frame.MissingFrame())
+    {
+        UpdateState(kPassive);
+    }
+    if (frame.MissingFrame() || !frame.Complete())
+    {
+        // State was corrupted, enable dual receiver.
+        UpdateState(kReceiving);
+    }
+}
+
+}

modules/video_coding/main/source/receiver.h

@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+
+#include "critical_section_wrapper.h"
+#include "jitter_buffer.h"
+#include "timing.h"
+#include "packet.h"
+
+namespace webrtc
+{
+
+class VCMEncodedFrame;
+
+enum VCMNackStatus
+{
+    kNackOk,
+    kNackNeedMoreMemory,
+    kNackKeyFrameRequest
+};
+
+enum VCMNackMode
+{
+    kNackInfinite,
+    kNoNack
+};
+
+enum VCMReceiverState
+{
+    kReceiving,
+    kPassive,
+    kWaitForPrimaryDecode
+};
+
+class VCMReceiver
+{
+public:
+    VCMReceiver(VCMTiming& timing,
+                WebRtc_Word32 vcmId = -1,
+                WebRtc_Word32 receiverId = -1,
+                bool master = true);
+    ~VCMReceiver();
+
+    WebRtc_Word32 Initialize();
+    void UpdateRtt(WebRtc_UWord32 rtt);
+    WebRtc_Word32 InsertPacket(const VCMPacket& packet,
+                               WebRtc_UWord16 frameWidth,
+                               WebRtc_UWord16 frameHeight);
+    VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+                                      WebRtc_Word64& nextRenderTimeMs,
+                                      bool renderTiming = true,
+                                      VCMReceiver* dualReceiver = NULL);
+    void ReleaseFrame(VCMEncodedFrame* frame);
+    WebRtc_Word32 ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate);
+    WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
+
+    // NACK
+    void SetNackMode(VCMNackMode nackMode);
+    VCMNackMode NackMode() const;
+    VCMNackStatus NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
+
+    // Dual decoder
+    bool DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const;
+    VCMReceiverState State() const;
+
+private:
+    VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
+                                      WebRtc_Word64 nextrenderTimeMs,
+                                      VCMReceiver* dualReceiver);
+    VCMEncodedFrame* FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
+                                       WebRtc_Word64 nextrenderTimeMs,
+                                       VCMReceiver* dualReceiver);
+    void CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver);
+    void UpdateState(VCMReceiverState newState);
+    void UpdateState(VCMEncodedFrame& frame);
+    static WebRtc_Word32 GenerateReceiverId();
+
+    CriticalSectionWrapper&    _critSect;
+    WebRtc_Word32           _vcmId;
+    WebRtc_Word32           _receiverId;
+    bool                    _master;
+    VCMJitterBuffer         _jitterBuffer;
+    VCMTiming&              _timing;
+    VCMEvent&               _renderWaitEvent;
+    VCMNackMode             _nackMode;
+    VCMReceiverState        _state;
+
+    static WebRtc_Word32    _receiverIdCounter;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_

modules/video_coding/main/source/rtt_filter.cc

@@ -0,0 +1,214 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "rtt_filter.h"
+
+#include <cmath>
+#include <stdlib.h>
+#include <string.h>
+
+namespace webrtc {
+
+VCMRttFilter::VCMRttFilter(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId)
+:
+_vcmId(vcmId),
+_receiverId(receiverId),
+_filtFactMax(35),
+_jumpStdDevs(2.5),
+_driftStdDevs(3.5),
+_detectThreshold(kMaxDriftJumpCount)
+{
+    Reset();
+}
+
+VCMRttFilter&
+VCMRttFilter::operator=(const VCMRttFilter& rhs)
+{
+    if (this != &rhs)
+    {
+        _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
+        _avgRtt = rhs._avgRtt;
+        _varRtt = rhs._varRtt;
+        _maxRtt = rhs._maxRtt;
+        _filtFactCount = rhs._filtFactCount;
+        _jumpCount = rhs._jumpCount;
+        _driftCount = rhs._driftCount;
+        memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
+        memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
+    }
+    return *this;
+}
+
+void
+VCMRttFilter::Reset()
+{
+    _gotNonZeroUpdate = false;
+    _avgRtt = 0;
+    _varRtt = 0;
+    _maxRtt = 0;
+    _filtFactCount = 1;
+    _jumpCount = 0;
+    _driftCount = 0;
+    memset(_jumpBuf, 0, kMaxDriftJumpCount);
+    memset(_driftBuf, 0, kMaxDriftJumpCount);
+}
+
+void
+VCMRttFilter::Update(WebRtc_UWord32 rttMs)
+{
+    if (!_gotNonZeroUpdate)
+    {
+        if (rttMs == 0)
+        {
+            return;
+        }
+        _gotNonZeroUpdate = true;
+    }
+
+    // Sanity check
+    if (rttMs > 3000)
+    {
+        rttMs = 3000;
+    }
+
+    double filtFactor = 0;
+    if (_filtFactCount > 1)
+    {
+        filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
+    }
+    _filtFactCount++;
+    if (_filtFactCount > _filtFactMax)
+    {
+        // This prevents filtFactor from going above
+        // (_filtFactMax - 1) / _filtFactMax,
+        // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
+        _filtFactCount = _filtFactMax;
+    }
+    double oldAvg = _avgRtt;
+    double oldVar = _varRtt;
+    _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
+    _varRtt = filtFactor * _varRtt + (1 - filtFactor) *
+                (rttMs - _avgRtt) * (rttMs - _avgRtt);
+    _maxRtt = VCM_MAX(rttMs, _maxRtt);
+    if (!JumpDetection(rttMs) || !DriftDetection(rttMs))
+    {
+        // In some cases we don't want to update the statistics
+        _avgRtt = oldAvg;
+        _varRtt = oldVar;
+    }
+    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+               "RttFilter Update: sample=%u avgRtt=%f varRtt=%f maxRtt=%u",
+               rttMs, _avgRtt, _varRtt, _maxRtt);
+}
+
+bool
+VCMRttFilter::JumpDetection(WebRtc_UWord32 rttMs)
+{
+    double diffFromAvg = _avgRtt - rttMs;
+    if (abs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
+    {
+        int diffSign = (diffFromAvg >= 0) ? 1 : -1;
+        int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
+        if (diffSign != jumpCountSign)
+        {
+            // Since the signs differ the samples currently
+            // in the buffer is useless as they represent a
+            // jump in a different direction.
+            _jumpCount = 0;
+        }
+        if (abs(_jumpCount) < kMaxDriftJumpCount)
+        {
+            // Update the buffer used for the short time
+            // statistics.
+            // The sign of the diff is used for updating the counter since
+            // we want to use the same buffer for keeping track of when
+            // the RTT jumps down and up.
+            _jumpBuf[abs(_jumpCount)] = rttMs;
+            _jumpCount += diffSign;
+        }
+        if (abs(_jumpCount) >= _detectThreshold)
+        {
+            // Detected an RTT jump
+            ShortRttFilter(_jumpBuf, abs(_jumpCount));
+            _filtFactCount = _detectThreshold + 1;
+            _jumpCount = 0;
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                       "Detected an RTT jump");
+        }
+        else
+        {
+            return false;
+        }
+    }
+    else
+    {
+        _jumpCount = 0;
+    }
+    return true;
+}
+
+bool
+VCMRttFilter::DriftDetection(WebRtc_UWord32 rttMs)
+{
+    if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt))
+    {
+        if (_driftCount < kMaxDriftJumpCount)
+        {
+            // Update the buffer used for the short time
+            // statistics.
+            _driftBuf[_driftCount] = rttMs;
+            _driftCount++;
+        }
+        if (_driftCount >= _detectThreshold)
+        {
+            // Detected an RTT drift
+            ShortRttFilter(_driftBuf, _driftCount);
+            _filtFactCount = _detectThreshold + 1;
+            _driftCount = 0;
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
+                       "Detected an RTT drift");
+        }
+    }
+    else
+    {
+        _driftCount = 0;
+    }
+    return true;
+}
+
+void
+VCMRttFilter::ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length)
+{
+    if (length == 0)
+    {
+        return;
+    }
+    _maxRtt = 0;
+    _avgRtt = 0;
+    for (WebRtc_UWord32 i=0; i < length; i++)
+    {
+        if (buf[i] > _maxRtt)
+        {
+            _maxRtt = buf[i];
+        }
+        _avgRtt += buf[i];
+    }
+    _avgRtt = _avgRtt / static_cast<double>(length);
+}
+
+WebRtc_UWord32
+VCMRttFilter::RttMs() const
+{
+    return static_cast<WebRtc_UWord32>(_maxRtt + 0.5);
+}
+
+}

modules/video_coding/main/source/rtt_filter.h

@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+class VCMRttFilter
+{
+public:
+    VCMRttFilter(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
+
+    VCMRttFilter& operator=(const VCMRttFilter& rhs);
+
+    // Resets the filter.
+    void Reset();
+    // Updates the filter with a new sample.
+    void Update(WebRtc_UWord32 rttMs);
+    // A getter function for the current RTT level in ms.
+    WebRtc_UWord32 RttMs() const;
+
+private:
+    // The size of the drift and jump memory buffers
+    // and thus also the detection threshold for these
+    // detectors in number of samples.
+    enum { kMaxDriftJumpCount = 5 };
+    // Detects RTT jumps by comparing the difference between
+    // samples and average to the standard deviation.
+    // Returns true if the long time statistics should be updated
+    // and false otherwise
+    bool JumpDetection(WebRtc_UWord32 rttMs);
+    // Detects RTT drifts by comparing the difference between
+    // max and average to the standard deviation.
+    // Returns true if the long time statistics should be updated
+    // and false otherwise
+    bool DriftDetection(WebRtc_UWord32 rttMs);
+    // Computes the short time average and maximum of the vector buf.
+    void ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length);
+
+    WebRtc_Word32         _vcmId;
+    WebRtc_Word32         _receiverId;
+    bool                  _gotNonZeroUpdate;
+    double                _avgRtt;
+    double                _varRtt;
+    WebRtc_UWord32        _maxRtt;
+    WebRtc_UWord32        _filtFactCount;
+    const WebRtc_UWord32  _filtFactMax;
+    const double          _jumpStdDevs;
+    const double          _driftStdDevs;
+    WebRtc_Word32         _jumpCount;
+    WebRtc_Word32         _driftCount;
+    const WebRtc_Word32   _detectThreshold;
+    WebRtc_UWord32        _jumpBuf[kMaxDriftJumpCount];
+    WebRtc_UWord32        _driftBuf[kMaxDriftJumpCount];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_

modules/video_coding/main/source/session_info.cc

@@ -0,0 +1,636 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "packet.h"
+#include "session_info.h"
+
+#include <string.h>
+#include <cassert>
+
+namespace webrtc {
+
+VCMSessionInfo::VCMSessionInfo():
+    _haveFirstPacket(false),
+    _markerBit(false),
+    _sessionNACK(false),
+    _completeSession(false),
+    _frameType(kVideoFrameDelta),
+    _previousFrameLoss(false),
+    _lowSeqNum(-1),
+    _highSeqNum(-1),
+    _highestPacketIndex(0)
+{
+    memset(_packetSizeBytes, 0, sizeof(_packetSizeBytes));
+    memset(_naluCompleteness, kNaluUnset, sizeof(_naluCompleteness));
+    memset(_ORwithPrevByte, 0, sizeof(_ORwithPrevByte));
+}
+
+VCMSessionInfo::~VCMSessionInfo()
+{
+}
+
+WebRtc_Word32 VCMSessionInfo::GetLowSeqNum() const
+{
+    return _lowSeqNum;
+}
+
+WebRtc_Word32 VCMSessionInfo::GetHighSeqNum() const
+{
+    return _highSeqNum;
+}
+
+void VCMSessionInfo::Reset()
+{
+    _lowSeqNum = -1;
+    _highSeqNum = -1;
+    _markerBit = false;
+    _haveFirstPacket = false;
+    _completeSession = false;
+    _frameType = kVideoFrameDelta;
+    _previousFrameLoss = false;
+    _sessionNACK = false;
+    _highestPacketIndex = 0;
+    memset(_packetSizeBytes, 0, sizeof(_packetSizeBytes));
+    memset(_naluCompleteness, kNaluUnset, sizeof(_naluCompleteness));
+    memset(_ORwithPrevByte, 0, sizeof(_ORwithPrevByte));
+}
+
+WebRtc_UWord32 VCMSessionInfo::GetSessionLength()
+{
+    WebRtc_UWord32 length = 0;
+    for (WebRtc_Word32 i=0; i<=_highestPacketIndex; ++i)
+    {
+        length += _packetSizeBytes[i];
+    }
+    return length;
+}
+
+void
+VCMSessionInfo::SetStartSeqNumber(WebRtc_UWord16 seqNumber)
+{
+    _lowSeqNum = seqNumber;
+    _highSeqNum = seqNumber;
+}
+
+bool
+VCMSessionInfo::HaveStartSeqNumber()
+{
+    if(_lowSeqNum == -1 || _highSeqNum == -1)
+    {
+        return false;
+    }
+    return true;
+}
+
+WebRtc_UWord32 VCMSessionInfo::InsertBuffer(WebRtc_UWord8* ptrStartOfLayer, WebRtc_Word32 packetIndex, const VCMPacket& packet)
+{
+    WebRtc_UWord32 moveLength = 0;
+    WebRtc_UWord32 returnLength = 0;
+    int i = 0;
+
+    // need to calc offset before updating _packetSizeBytes
+    WebRtc_UWord32 offset = 0;
+    WebRtc_UWord32 packetSize = 0;
+
+    // Store this packet length. Add length since we could have data present already (e.g. multicall case).
+    if (packet.bits)
+    {
+        packetSize = packet.sizeBytes;
+    }
+    else
+    {
+        packetSize = packet.sizeBytes + (packet.insertStartCode?kH264StartCodeLengthBytes:0);
+    }
+
+    _packetSizeBytes[packetIndex] += packetSize;
+
+    // count only the one in our layer
+    for (i=0; i<packetIndex; ++i)
+    {
+        offset += _packetSizeBytes[i];
+    }
+    for (i=packetIndex+1; i<=_highestPacketIndex; ++i)
+    {
+        moveLength += _packetSizeBytes[i];
+    }
+    if (moveLength > 0)
+    {
+        memmove((void*)(ptrStartOfLayer + offset + packetSize), ptrStartOfLayer + offset, moveLength);
+    }
+
+    if (packet.bits)
+    {
+        // Add the packet without ORing end and start bytes together.
+        // This is done when the frame is fetched for decoding by calling
+        // GlueTogether().
+        _ORwithPrevByte[packetIndex] = true;
+        if (packet.dataPtr != NULL)
+        {
+            memcpy((void*)(ptrStartOfLayer + offset), packet.dataPtr, packetSize);
+        }
+        returnLength = packetSize;
+    }
+    else
+    {
+        _ORwithPrevByte[packetIndex] = false;
+        if (packet.dataPtr != NULL)
+        {
+            const unsigned char startCode[]      = {0, 0, 0, 1};
+            if(packet.insertStartCode)
+            {
+                memcpy((void*)(ptrStartOfLayer + offset), startCode, kH264StartCodeLengthBytes);
+            }
+            memcpy((void*)(ptrStartOfLayer + offset
+                + (packet.insertStartCode?kH264StartCodeLengthBytes:0)),
+                packet.dataPtr,
+                packet.sizeBytes);
+        }
+        returnLength = packetSize;
+    }
+
+    if (packet.isFirstPacket)
+    {
+        _haveFirstPacket = true;
+    }
+    if (packet.markerBit)
+    {
+        _markerBit = true;
+    }
+     // Store information about if the packet is decodable as is or not.
+    _naluCompleteness[packetIndex]=packet.completeNALU;
+
+    UpdateCompleteSession();
+
+    return returnLength;
+}
+
+void VCMSessionInfo::UpdateCompleteSession()
+{
+    if (_haveFirstPacket && _markerBit)
+    {
+        // do we have all packets in this session?
+        bool completeSession = true;
+        for (int i=0; i<= _highestPacketIndex; ++i)
+        {
+            if (_naluCompleteness[i] == kNaluUnset)
+            {
+                completeSession = false;
+                break;
+            }
+        }
+        _completeSession = completeSession;
+    }
+}
+
+bool VCMSessionInfo::IsSessionComplete()
+{
+    return _completeSession;
+}
+
+
+// Find the start and end index of packetIndex packet.
+// startIndex -1 if start not found endIndex=-1 if end index not found
+void VCMSessionInfo::FindNaluBorder(WebRtc_Word32 packetIndex,WebRtc_Word32& startIndex, WebRtc_Word32& endIndex)
+{
+
+        if(_naluCompleteness[packetIndex]==kNaluStart ||
+             _naluCompleteness[packetIndex]==kNaluComplete)
+        {
+            startIndex=packetIndex;
+        }
+        else // Need to find the start
+        {
+            for(startIndex=packetIndex-1;startIndex>=0;--startIndex)
+            {
+
+                if( (_naluCompleteness[startIndex]==kNaluComplete && _packetSizeBytes[startIndex]>0) ||(_naluCompleteness[startIndex]==kNaluEnd && startIndex>0)) // Found previous NALU.
+                {
+                    startIndex++;
+                    break;
+                }
+                if( _naluCompleteness[startIndex]==kNaluStart) // This is where the NALU start.
+                {
+                    break;
+                }
+            }
+        }
+
+        if(_naluCompleteness[packetIndex]==kNaluEnd ||
+             _naluCompleteness[packetIndex]==kNaluComplete)
+        {
+            endIndex=packetIndex;
+        }
+        else
+        {
+            // Find the next NALU
+            for(endIndex=packetIndex+1;endIndex<=_highestPacketIndex;++endIndex)
+            {
+                if((_naluCompleteness[endIndex]==kNaluComplete && _packetSizeBytes[endIndex]>0) || _naluCompleteness[endIndex]==kNaluStart) // Found next NALU.
+                {
+                    endIndex--;
+                    break;
+                }
+                if( _naluCompleteness[endIndex]==kNaluEnd) // This is where the NALU end.
+                {
+                    break;
+                }
+            }
+            if(endIndex>_highestPacketIndex)
+                endIndex=-1;
+        }
+}
+
+// Deletes all packets between startIndex and endIndex
+WebRtc_UWord32 VCMSessionInfo::DeletePackets(WebRtc_UWord8* ptrStartOfLayer,WebRtc_Word32 startIndex,WebRtc_Word32 endIndex)
+{
+
+    //Get the number of bytes to delete.
+    //Clear the size of these packets.
+    WebRtc_UWord32 bytesToDelete=0; /// The number of bytes to delete.
+    for(int j=startIndex;j<=endIndex;++j)
+    {
+        bytesToDelete+=_packetSizeBytes[j];
+        _packetSizeBytes[j]=0;
+    }
+    if (bytesToDelete > 0)
+    {
+        // Get the offset we want to move to.
+        int destOffset=0;
+        for(int j=0;j<startIndex;j++)
+        {
+           destOffset+=_packetSizeBytes[j];
+        }
+
+        //Get the number of bytes to move
+        WebRtc_UWord32 numberOfBytesToMove=0;
+        for (int j=endIndex+1; j<=_highestPacketIndex; ++j)
+        {
+            numberOfBytesToMove += _packetSizeBytes[j];
+        }
+
+        memmove((void*)(ptrStartOfLayer + destOffset),(void*)(ptrStartOfLayer + destOffset+bytesToDelete), numberOfBytesToMove);
+
+    }
+
+    return bytesToDelete;
+}
+
+// Makes the layer decodable. Ie only contain decodable NALU
+// return the number of bytes deleted from the session. -1 if an error occurs
+WebRtc_UWord32 VCMSessionInfo::MakeSessionDecodable(WebRtc_UWord8* ptrStartOfLayer)
+{
+    if(_lowSeqNum<0) // No packets in this session
+        return 0;
+
+    WebRtc_Word32 startIndex=0;
+    WebRtc_Word32 endIndex=0;
+    int packetIndex=0;
+    WebRtc_UWord32 returnLength=0;
+    for (packetIndex=0; packetIndex<= _highestPacketIndex; ++packetIndex)
+    {
+        if (_naluCompleteness[packetIndex] == kNaluUnset) // Found a lost packet
+        {
+            FindNaluBorder(packetIndex,startIndex,endIndex);
+            if(startIndex==-1)
+                startIndex=0;
+            if(endIndex==-1)
+                endIndex=_highestPacketIndex;
+
+            returnLength+=DeletePackets(ptrStartOfLayer,packetIndex,endIndex);
+            packetIndex=endIndex;
+        }// end lost packet
+    }
+
+    //Make sure the first packet is decodable (Either complete nalu or start of NALU)
+    if(_packetSizeBytes[0]>0)
+    {
+        switch(_naluCompleteness[0])
+        {
+            case kNaluComplete: //Packet can be decoded as is.
+                break;
+
+            case kNaluStart: // Packet contain beginning of NALU- No need to do anything.
+                break;
+            case kNaluIncomplete: //Packet is not beginning or end of NALU
+                //Need to find the end of this fua NALU and delete all packets.
+                FindNaluBorder(0,startIndex,endIndex);
+                if(endIndex==-1) // No end found. Delete
+                {
+                    endIndex=_highestPacketIndex;
+                }
+                returnLength+=DeletePackets(ptrStartOfLayer,0,endIndex);//Delete this NALU.
+                break;
+            case kNaluEnd:    // Packet is the end of a NALU
+                //Need to delete this packet
+                returnLength+=DeletePackets(ptrStartOfLayer,0,0);//Delete this NALU.
+                break;
+            default:
+                assert(false);
+        }
+    }
+
+    return returnLength;
+}
+
+WebRtc_Word32 VCMSessionInfo::ZeroOutSeqNum(WebRtc_Word32* list, WebRtc_Word32 num)
+{
+    if ((NULL == list) || (num < 1))
+    {
+        return -1;
+    }
+    if (_lowSeqNum == -1)
+    {
+        // no packets in this frame
+        return 0;
+    }
+
+    // Find end point (index of entry that equals _lowSeqNum)
+    int index = 0;
+    for (; index <num; index++)
+    {
+        if (list[index] == _lowSeqNum)
+        {
+            list[index] = -1;
+            break;
+        }
+    }
+
+    // Zero out between first entry and end point
+    int i = 0;
+    while ( i <= _highestPacketIndex && index < num)
+    {
+        if (_naluCompleteness[i] != kNaluUnset)
+        {
+            list[index] = -1;
+        }
+        else
+        {
+            _sessionNACK = true;
+        }
+        i++;
+        index++;
+    }
+    if(!_haveFirstPacket)
+    {
+        _sessionNACK = true;
+    }
+    return 0;
+}
+
+WebRtc_Word32 VCMSessionInfo::GetHighestPacketIndex()
+{
+    return _highestPacketIndex;
+}
+
+bool VCMSessionInfo::HaveLastPacket()
+{
+    return _markerBit;
+}
+
+void VCMSessionInfo::ForceSetHaveLastPacket()
+{
+    _markerBit = true;
+    UpdateCompleteSession();
+}
+
+bool VCMSessionInfo::IsRetransmitted()
+{
+    return _sessionNACK;
+}
+
+void VCMSessionInfo::UpdatePacketSize(WebRtc_Word32 packetIndex, WebRtc_UWord32 length)
+{
+    // sanity
+    if(packetIndex >= kMaxPacketsInJitterBuffer || packetIndex < 0)
+    {
+        //not allowed
+        assert(!"SessionInfo::UpdatePacketSize Error: invalid packetIndex");
+        return;
+    }
+    _packetSizeBytes[packetIndex] = length;
+}
+
+void VCMSessionInfo::PrependPacketIndices(WebRtc_Word32 numberOfPacketIndices)
+{
+    // sanity
+    if((numberOfPacketIndices + GetHighestPacketIndex() >= kMaxPacketsInJitterBuffer) || numberOfPacketIndices < 0)
+    {
+        //not allowed
+        assert(!"SessionInfo::PrependPacketIndexes Error: invalid packetIndex");
+        return;
+    }
+    // Works if we have new packets before packetIndex = 0
+    int numOfPacketsToMove = GetHighestPacketIndex()+1;
+    memmove(&_packetSizeBytes[numberOfPacketIndices], &_packetSizeBytes[0], (numOfPacketsToMove)*sizeof(WebRtc_UWord16));
+    memset(&_packetSizeBytes[0], 0, numberOfPacketIndices*sizeof(WebRtc_UWord16));
+
+    _highestPacketIndex += (WebRtc_UWord16)numberOfPacketIndices;
+}
+
+void VCMSessionInfo::ClearPacketSize(WebRtc_Word32 packetIndex)
+{
+    // sanity
+    if(packetIndex >= kMaxPacketsInJitterBuffer || packetIndex < 0)
+    {
+        //not allowed
+        assert(!"SessionInfo::ClearPacketSize Error: invalid packetIndex");
+        return;
+    }
+    _packetSizeBytes[packetIndex] =0;
+}
+
+WebRtc_UWord32 VCMSessionInfo::GetPacketSize(WebRtc_Word32 packetIndex)
+{
+    // sanity
+    if(packetIndex >= kMaxPacketsInJitterBuffer || packetIndex < 0)
+    {
+        //not allowed
+        assert(!"SessionInfo::GetPacketSize Error: invalid packetIndex");
+        return 0;
+    }
+    return _packetSizeBytes[packetIndex];
+}
+
+WebRtc_Word64
+VCMSessionInfo::InsertPacket(const VCMPacket& packet, WebRtc_UWord8* ptrStartOfLayer)
+{
+    //not allowed
+    assert(!packet.insertStartCode || !packet.bits);
+
+    // Check if this is first packet (only valid for some codecs)
+    if (packet.isFirstPacket)
+    {
+        // the first packet in the frame always signals the frametype
+        _frameType = packet.frameType;
+    }
+
+    // Check sequence number and update highest and lowest sequence numbers received.
+    // Move data if this seq num is lower than previously lowest.
+
+    if (packet.seqNum > _highSeqNum)
+    {
+        // This packet's seq num is higher than previously highest seq num; normal case
+        // if we have a wrap, only update with wrapped values
+        if (!(_highSeqNum < 0x00ff && packet.seqNum > 0xff00))
+        {
+            _highSeqNum = packet.seqNum;
+        }
+    } else if (_highSeqNum > 0xff00 && packet.seqNum < 0x00ff)
+    {
+        // wrap
+        _highSeqNum = packet.seqNum;
+    }
+    int packetIndex = packet.seqNum - (WebRtc_UWord16)_lowSeqNum;
+    if(_lowSeqNum < 0x00ff && packet.seqNum > 0xff00)
+    {
+        // negative wrap
+        packetIndex = packet.seqNum - 0x10000 - _lowSeqNum;
+    }
+    if (packetIndex < 0)
+    {
+        if (_lowSeqNum > 0xff00 && packet.seqNum < 0x00ff)
+        {
+            // we have a false detect due to the wrap
+            packetIndex = (0xffff - (WebRtc_UWord16)_lowSeqNum) + packet.seqNum + (WebRtc_UWord16)1;
+        } else
+        {
+            // This packet's seq num is lower than previously lowest seq num, but no wrap
+            // We need to move the data in all arrays indexed by packetIndex and insert the new
+            // packet's info
+            // How many packets should we leave room for (positions to shift)?
+            // Example - this seq num is 3 lower than previously lowest seq num
+            // Before: |--prev packet with lowest seq num--|--|...|
+            // After:  |--new lowest seq num--|--|--|--prev packet with lowest seq num--|--|...|
+
+            WebRtc_UWord16 positionsToShift   = (WebRtc_UWord16)_lowSeqNum - packet.seqNum;
+            WebRtc_UWord16 numOfPacketsToMove = _highestPacketIndex + 1;
+
+            // sanity, do we have room for the shift?
+            if ((positionsToShift + numOfPacketsToMove) > kMaxPacketsInJitterBuffer)
+            {
+                return -1;
+            }
+
+            // Shift _ORwithPrevByte array
+            memmove(&_ORwithPrevByte[positionsToShift],
+                &_ORwithPrevByte[0], numOfPacketsToMove*sizeof(bool));
+            memset(&_ORwithPrevByte[0], false, positionsToShift*sizeof(bool));
+
+            // Shift _packetSizeBytes array
+            memmove(&_packetSizeBytes[positionsToShift],
+                &_packetSizeBytes[0], numOfPacketsToMove*sizeof(WebRtc_UWord32));
+            memset(&_packetSizeBytes[0], 0, positionsToShift*sizeof(WebRtc_UWord32));
+
+            //Shift _naluCompleteness
+            memmove(&_naluCompleteness[positionsToShift],
+                   &_naluCompleteness[0], numOfPacketsToMove*sizeof(WebRtc_UWord8));
+            memset(&_naluCompleteness[0], kNaluUnset, positionsToShift*sizeof(WebRtc_UWord8));
+
+            _highestPacketIndex += positionsToShift;
+            _lowSeqNum = packet.seqNum;
+            packetIndex = 0; // (seqNum - _lowSeqNum) = 0
+        }
+    } // if (_lowSeqNum > seqNum)
+
+    // sanity
+    if (packetIndex >= kMaxPacketsInJitterBuffer )
+    {
+        return -1;
+    }
+    if (packetIndex < 0 )
+    {
+        return -1;
+    }
+
+    // Check for duplicate packets
+    if (_packetSizeBytes[packetIndex] != 0)
+    {
+        // We have already received a packet with this sequence number, ignore it.
+        return -2;
+    }
+
+    // update highest packet index
+    _highestPacketIndex = packetIndex > _highestPacketIndex ? packetIndex :_highestPacketIndex;
+
+    return InsertBuffer(ptrStartOfLayer, packetIndex, packet);
+}
+
+WebRtc_UWord32 VCMSessionInfo::PrepareForDecode(WebRtc_UWord8* ptrStartOfLayer, VideoCodecType codec)
+{
+    WebRtc_UWord32 currentPacketOffset = 0;
+    WebRtc_UWord32 length = GetSessionLength();
+    WebRtc_UWord32 idSum = 0;
+    WebRtc_UWord32 realDataBytes = 0;
+    if (length == 0)
+    {
+        return length;
+    }
+    bool previousLost = false;
+    for (int i=0; i <= _highestPacketIndex; i++)
+    {
+        if (_ORwithPrevByte[i])
+        {
+            if (currentPacketOffset > 0)
+            {
+                WebRtc_UWord8* ptrFirstByte = ptrStartOfLayer + currentPacketOffset;
+
+                if (_packetSizeBytes[i-1] == 0 || previousLost)
+                {
+                    // It is be better to throw away this packet if we are missing the
+                    // previous packet.
+                    memset(ptrFirstByte, 0, _packetSizeBytes[i]);
+                    previousLost = true;
+                }
+                else if (_packetSizeBytes[i] > 0) // Ignore if empty packet
+                {
+                    // Glue with previous byte
+                    // Move everything from [this packet start + 1, end of buffer] one byte to the left
+                    WebRtc_UWord8* ptrPrevByte = ptrFirstByte - 1;
+                    *ptrPrevByte = (*ptrPrevByte) | (*ptrFirstByte);
+                    WebRtc_UWord32 lengthToEnd = length - (currentPacketOffset + 1);
+                    memmove((void*)ptrFirstByte, (void*)(ptrFirstByte + 1), lengthToEnd);
+                    _packetSizeBytes[i]--;
+                    length--;
+                    previousLost = false;
+                    realDataBytes += _packetSizeBytes[i];
+                }
+            }
+            else
+            {
+                memset(ptrStartOfLayer, 0, _packetSizeBytes[i]);
+                previousLost = true;
+            }
+        }
+        else if (_packetSizeBytes[i] == 0 && codec == kVideoCodecH263)
+        {
+            WebRtc_UWord8* ptrFirstByte = ptrStartOfLayer + currentPacketOffset;
+            memmove(ptrFirstByte + 10, ptrFirstByte, length - currentPacketOffset);
+            memset(ptrFirstByte, 0, 10);
+            _packetSizeBytes[i] = 10;
+            length += _packetSizeBytes[i];
+            previousLost = true;
+        }
+        else
+        {
+            realDataBytes += _packetSizeBytes[i];
+            previousLost = false;
+        }
+        currentPacketOffset += _packetSizeBytes[i];
+    }
+    if (realDataBytes == 0)
+    {
+        // Drop the frame since all it contains are zeros
+        length = 0;
+        memset(_packetSizeBytes, 0, sizeof(_packetSizeBytes));
+    }
+    return length;
+}
+
+}

modules/video_coding/main/source/session_info.h

@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+#include "packet.h"
+
+namespace webrtc
+{
+
+class VCMSessionInfo
+{
+public:
+    VCMSessionInfo();
+    virtual ~VCMSessionInfo();
+
+    VCMSessionInfo(const VCMSessionInfo& rhs);
+
+    WebRtc_Word32 ZeroOutSeqNum(WebRtc_Word32* list, WebRtc_Word32 num);
+    virtual void Reset();
+
+    WebRtc_Word64 InsertPacket(const VCMPacket& packet, WebRtc_UWord8* ptrStartOfLayer);
+
+    virtual bool IsSessionComplete();
+    WebRtc_UWord32 MakeSessionDecodable(WebRtc_UWord8* ptrStartOfLayer);
+
+    WebRtc_UWord32 GetSessionLength();
+    bool HaveLastPacket();
+    void ForceSetHaveLastPacket();
+    bool IsRetransmitted();
+    webrtc::FrameType FrameType() const { return _frameType; }
+
+    virtual WebRtc_Word32 GetHighestPacketIndex();
+    virtual WebRtc_UWord32 GetPacketSize(WebRtc_Word32 packetIndex);
+    virtual void ClearPacketSize(WebRtc_Word32 packetIndex);
+    virtual void UpdatePacketSize(WebRtc_Word32 packetIndex, WebRtc_UWord32 length);
+    virtual void PrependPacketIndices(WebRtc_Word32 numberOfPacketIndexes);
+
+    void SetStartSeqNumber(WebRtc_UWord16 seqNumber);
+
+    bool HaveStartSeqNumber();
+
+    WebRtc_Word32 GetLowSeqNum() const;
+    WebRtc_Word32 GetHighSeqNum() const;
+
+    WebRtc_UWord32 PrepareForDecode(WebRtc_UWord8* ptrStartOfLayer, VideoCodecType codec);
+
+    void SetPreviousFrameLoss() { _previousFrameLoss = true; }
+    bool PreviousFrameLoss() const { return _previousFrameLoss; }
+
+protected:
+    WebRtc_UWord32 InsertBuffer(WebRtc_UWord8* ptrStartOfLayer,
+                                WebRtc_Word32 packetIndex,
+                                const VCMPacket& packet);
+    void FindNaluBorder(WebRtc_Word32 packetIndex,
+                        WebRtc_Word32& startIndex,
+                        WebRtc_Word32& endIndex);
+    WebRtc_UWord32 DeletePackets(WebRtc_UWord8* ptrStartOfLayer,
+                                 WebRtc_Word32 startIndex,
+                                 WebRtc_Word32 endIndex);
+    void UpdateCompleteSession();
+
+    bool _haveFirstPacket;      // If we have inserted the first packet into this frame
+    bool _markerBit;            // If we have inserted a packet with markerbit into this frame
+    bool _sessionNACK;          // If this session has been NACKed by JB
+    bool _completeSession;
+    webrtc::FrameType  _frameType;
+    bool           _previousFrameLoss;
+
+    WebRtc_Word32  _lowSeqNum;          // Lowest packet sequence number in a session
+    WebRtc_Word32  _highSeqNum;         // Highest packet sequence number in a session
+
+    // Highest packet index in this frame
+    WebRtc_UWord16 _highestPacketIndex;
+    // Length of packet (used for reordering)
+    WebRtc_UWord32 _packetSizeBytes[kMaxPacketsInJitterBuffer];
+    // Completness of packets. Used for deciding if the frame is decodable.
+    WebRtc_UWord8  _naluCompleteness[kMaxPacketsInJitterBuffer];
+    bool           _ORwithPrevByte[kMaxPacketsInJitterBuffer];
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_

modules/video_coding/main/source/tick_time.h

@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_
+
+#include "tick_util.h"
+
+#include <assert.h>
+
+namespace webrtc
+{
+
+//#define TICK_TIME_DEBUG
+
+class VCMTickTime : public TickTime
+{
+#ifdef TICK_TIME_DEBUG
+public:
+    /*
+    *   Get current time
+    */
+    static TickTime Now() { assert(false); };
+
+    /*
+    *   Get time in milli seconds
+    */
+    static WebRtc_Word64 MillisecondTimestamp() { return _timeNowDebug; };
+
+    /*
+    *   Get time in micro seconds
+    */
+    static WebRtc_Word64 MicrosecondTimestamp() { return _timeNowDebug * 1000LL; };
+
+    static void IncrementDebugClock() { _timeNowDebug++; };
+
+private:
+    static WebRtc_Word64     _timeNowDebug;
+
+#else
+public:
+    static void IncrementDebugClock() { assert(false); };
+#endif
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TICK_TIME_H_

modules/video_coding/main/source/timestamp_extrapolator.cc

@@ -0,0 +1,259 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "internal_defines.h"
+#include "timestamp_extrapolator.h"
+#include "tick_time.h"
+#include "trace.h"
+
+namespace webrtc {
+
+VCMTimestampExtrapolator::VCMTimestampExtrapolator(WebRtc_Word32 vcmId, WebRtc_Word32 id)
+:
+_rwLock(*RWLockWrapper::CreateRWLock()),
+_vcmId(vcmId),
+_id(id),
+_startMs(0),
+_firstTimestamp(0),
+_wrapArounds(0),
+_prevTs90khz(0),
+_lambda(1),
+_firstAfterReset(true),
+_packetCount(0),
+_startUpFilterDelayInPackets(2),
+_detectorAccumulatorPos(0),
+_detectorAccumulatorNeg(0),
+_alarmThreshold(60e3),
+_accDrift(6600), // in timestamp ticks, i.e. 15 ms
+_accMaxError(7000),
+_P11(1e10)
+{
+    Reset(VCMTickTime::MillisecondTimestamp());
+}
+
+VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
+{
+    delete &_rwLock;
+}
+
+void
+VCMTimestampExtrapolator::Reset(const WebRtc_Word64 nowMs /* = -1 */)
+{
+    WriteLockScoped wl(_rwLock);
+    if (nowMs > -1)
+    {
+        _startMs = nowMs;
+    }
+    else
+    {
+        _startMs = VCMTickTime::MillisecondTimestamp();
+    }
+    _prevMs = _startMs;
+    _firstTimestamp = 0;
+    _w[0] = 90.0;
+    _w[1] = 0;
+    _P[0][0] = 1;
+    _P[1][1] = _P11;
+    _P[0][1] = _P[1][0] = 0;
+    _firstAfterReset = true;
+    _prevTs90khz = 0;
+    _wrapArounds = 0;
+    _packetCount = 0;
+    _detectorAccumulatorPos = 0;
+    _detectorAccumulatorNeg = 0;
+}
+
+void
+VCMTimestampExtrapolator::Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace)
+{
+
+    _rwLock.AcquireLockExclusive();
+    if (tMs - _prevMs > 10e3)
+    {
+        // Ten seconds without a complete frame.
+        // Reset the extrapolator
+        _rwLock.ReleaseLockExclusive();
+        Reset();
+        _rwLock.AcquireLockExclusive();
+    }
+    else
+    {
+        _prevMs = tMs;
+    }
+
+    // Remove offset to prevent badly scaled matrices
+    tMs -= _startMs;
+
+    WebRtc_Word32 prevWrapArounds = _wrapArounds;
+    CheckForWrapArounds(ts90khz);
+    WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+    if (wrapAroundsSincePrev == 0 && ts90khz < _prevTs90khz)
+    {
+        _rwLock.ReleaseLockExclusive();
+        return;
+    }
+
+    if (_firstAfterReset)
+    {
+        // Make an initial guess of the offset,
+        // should be almost correct since tMs - _startMs
+        // should about zero at this time.
+        _w[1] = -_w[0] * tMs;
+        _firstTimestamp = ts90khz;
+        _firstAfterReset = false;
+    }
+
+    // Compensate for wraparounds by changing the line offset
+    _w[1] = _w[1] - wrapAroundsSincePrev * ((static_cast<WebRtc_Word64>(1)<<32) - 1);
+
+    double residual = (static_cast<double>(ts90khz) - _firstTimestamp) - static_cast<double>(tMs) * _w[0] - _w[1];
+    if (DelayChangeDetection(residual, trace) &&
+        _packetCount >= _startUpFilterDelayInPackets)
+    {
+        // A sudden change of average network delay has been detected.
+        // Force the filter to adjust its offset parameter by changing
+        // the offset uncertainty. Don't do this during startup.
+        _P[1][1] = _P11;
+    }
+    //T = [t(k) 1]';
+    //that = T'*w;
+    //K = P*T/(lambda + T'*P*T);
+    double K[2];
+    K[0] = _P[0][0] * tMs + _P[0][1];
+    K[1] = _P[1][0] * tMs + _P[1][1];
+    double TPT = _lambda + tMs * K[0] + K[1];
+    K[0] /= TPT;
+    K[1] /= TPT;
+    //w = w + K*(ts(k) - that);
+    _w[0] = _w[0] + K[0] * residual;
+    _w[1] = _w[1] + K[1] * residual;
+    //P = 1/lambda*(P - K*T'*P);
+    double p00 = 1 / _lambda * (_P[0][0] - (K[0] * tMs * _P[0][0] + K[0] * _P[1][0]));
+    double p01 = 1 / _lambda * (_P[0][1] - (K[0] * tMs * _P[0][1] + K[0] * _P[1][1]));
+    _P[1][0] = 1 / _lambda * (_P[1][0] - (K[1] * tMs * _P[0][0] + K[1] * _P[1][0]));
+    _P[1][1] = 1 / _lambda * (_P[1][1] - (K[1] * tMs * _P[0][1] + K[1] * _P[1][1]));
+    _P[0][0] = p00;
+    _P[0][1] = p01;
+    if (_packetCount < _startUpFilterDelayInPackets)
+    {
+        _packetCount++;
+    }
+    if (trace)
+    {
+        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id),  "w[0]=%f w[1]=%f ts=%u tMs=%u", _w[0], _w[1], ts90khz, tMs);
+    }
+    _rwLock.ReleaseLockExclusive();
+}
+
+WebRtc_UWord32
+VCMTimestampExtrapolator::ExtrapolateTimestamp(WebRtc_Word64 tMs) const
+{
+    ReadLockScoped rl(_rwLock);
+    WebRtc_UWord32 timestamp = 0;
+    if (_packetCount == 0)
+    {
+        timestamp = 0;
+    }
+    else if (_packetCount < _startUpFilterDelayInPackets)
+    {
+        timestamp = static_cast<WebRtc_UWord32>(90.0 * (tMs - _prevMs) + _prevTs90khz + 0.5);
+    }
+    else
+    {
+        timestamp = static_cast<WebRtc_UWord32>(_w[0] * (tMs - _startMs) + _w[1] + _firstTimestamp + 0.5);
+    }
+    return timestamp;
+}
+
+WebRtc_Word64
+VCMTimestampExtrapolator::ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const
+{
+    ReadLockScoped rl(_rwLock);
+    WebRtc_Word64 localTimeMs = 0;
+    if (_packetCount == 0)
+    {
+        localTimeMs = -1;
+    }
+    else if (_packetCount < _startUpFilterDelayInPackets)
+    {
+        localTimeMs = _prevMs + static_cast<WebRtc_Word64>(static_cast<double>(timestamp90khz - _prevTs90khz) / 90.0 + 0.5);
+    }
+    else
+    {
+        if (_w[0] < 1e-3)
+        {
+            localTimeMs = _startMs;
+        }
+        else
+        {
+            double timestampDiff = static_cast<double>(timestamp90khz) - static_cast<double>(_firstTimestamp);
+            localTimeMs = static_cast<WebRtc_Word64>(static_cast<double>(_startMs) + (timestampDiff - _w[1]) / _w[0] + 0.5);
+        }
+    }
+    return localTimeMs;
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp and
+// keeps track of the number of wrap arounds since reset.
+void
+VCMTimestampExtrapolator::CheckForWrapArounds(WebRtc_UWord32 ts90khz)
+{
+    if (_prevTs90khz == 0)
+    {
+        _prevTs90khz = ts90khz;
+        return;
+    }
+    if (ts90khz < _prevTs90khz)
+    {
+        // This difference will probably be less than -2^31 if we have had a wrap around
+        // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is casted to a Word32,
+        // it should be positive.
+        if (static_cast<WebRtc_Word32>(ts90khz - _prevTs90khz) > 0)
+        {
+            // Forward wrap around
+            _wrapArounds++;
+        }
+    }
+    // This difference will probably be less than -2^31 if we have had a backward wrap around.
+    // Since it is casted to a Word32, it should be positive.
+    else if (static_cast<WebRtc_Word32>(_prevTs90khz - ts90khz) > 0)
+    {
+        // Backward wrap around
+        _wrapArounds--;
+    }
+    _prevTs90khz = ts90khz;
+}
+
+bool
+VCMTimestampExtrapolator::DelayChangeDetection(double error, bool trace)
+{
+    // CUSUM detection of sudden delay changes
+    error = (error > 0) ? VCM_MIN(error, _accMaxError) : VCM_MAX(error, -_accMaxError);
+    _detectorAccumulatorPos = VCM_MAX(_detectorAccumulatorPos + error - _accDrift, (double)0);
+    _detectorAccumulatorNeg = VCM_MIN(_detectorAccumulatorNeg + error + _accDrift, (double)0);
+    if (_detectorAccumulatorPos > _alarmThreshold || _detectorAccumulatorNeg < -_alarmThreshold)
+    {
+        // Alarm
+        if (trace)
+        {
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id),  "g1=%f g2=%f alarm=1", _detectorAccumulatorPos, _detectorAccumulatorNeg);
+        }
+        _detectorAccumulatorPos = _detectorAccumulatorNeg = 0;
+        return true;
+    }
+    if (trace)
+    {
+        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id),  "g1=%f g2=%f alarm=0", _detectorAccumulatorPos, _detectorAccumulatorNeg);
+    }
+    return false;
+}
+
+}

modules/video_coding/main/source/timestamp_extrapolator.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
+
+#include "typedefs.h"
+#include "rw_lock_wrapper.h"
+
+namespace webrtc
+{
+
+class VCMTimestampExtrapolator
+{
+public:
+    VCMTimestampExtrapolator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
+    ~VCMTimestampExtrapolator();
+    void Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace = true);
+    WebRtc_UWord32 ExtrapolateTimestamp(WebRtc_Word64 tMs) const;
+    WebRtc_Word64 ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const;
+    void Reset(WebRtc_Word64 nowMs = -1);
+
+private:
+    void CheckForWrapArounds(WebRtc_UWord32 ts90khz);
+    bool DelayChangeDetection(double error, bool trace = true);
+    RWLockWrapper&         _rwLock;
+    WebRtc_Word32         _vcmId;
+    WebRtc_Word32         _id;
+    bool                _trace;
+    double              _w[2];
+    double              _P[2][2];
+    WebRtc_Word64         _startMs;
+    WebRtc_Word64         _prevMs;
+    WebRtc_UWord32        _firstTimestamp;
+    WebRtc_Word32         _wrapArounds;
+    WebRtc_UWord32        _prevTs90khz;
+    const double        _lambda;
+    bool                _firstAfterReset;
+    WebRtc_UWord32        _packetCount;
+    const WebRtc_UWord32  _startUpFilterDelayInPackets;
+
+    double              _detectorAccumulatorPos;
+    double              _detectorAccumulatorNeg;
+    const double        _alarmThreshold;
+    const double        _accDrift;
+    const double        _accMaxError;
+    const double        _P11;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_

modules/video_coding/main/source/timestamp_map.cc

@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "timestamp_map.h"
+#include <stdlib.h>
+#include <assert.h>
+
+namespace webrtc {
+
+// Constructor. Optional parameter specifies maximum number of
+// coexisting timers.
+VCMTimestampMap::VCMTimestampMap(WebRtc_Word32 length):
+    _nextAddIx(0),
+    _nextPopIx(0)
+{
+    if (length <= 0)
+    {
+        // default
+        length = 10;
+    }
+
+    _map = new VCMTimestampDataTuple[length];
+    _length = length;
+}
+
+// Destructor.
+VCMTimestampMap::~VCMTimestampMap()
+{
+    delete [] _map;
+}
+
+// Empty the list of timers.
+void
+VCMTimestampMap::Reset()
+{
+    _nextAddIx = 0;
+    _nextPopIx = 0;
+}
+
+WebRtc_Word32
+VCMTimestampMap::Add(WebRtc_UWord32 timestamp, void* data)
+{
+    _map[_nextAddIx].timestamp = timestamp;
+    _map[_nextAddIx].data = data;
+    _nextAddIx = (_nextAddIx + 1) % _length;
+
+    if (_nextAddIx == _nextPopIx)
+    {
+        // Circular list full; forget oldest entry
+        _nextPopIx = (_nextPopIx + 1) % _length;
+        return -1;
+    }
+    return 0;
+}
+
+void*
+VCMTimestampMap::Pop(WebRtc_UWord32 timestamp)
+{
+    while (!IsEmpty())
+    {
+        if (_map[_nextPopIx].timestamp == timestamp)
+        {
+            // found start time for this timestamp
+            void* data = _map[_nextPopIx].data;
+            _map[_nextPopIx].data = NULL;
+            _nextPopIx = (_nextPopIx + 1) % _length;
+            return data;
+        }
+        else if (_map[_nextPopIx].timestamp > timestamp)
+        {
+            // the timestamp we are looking for is not in the list
+            assert(_nextPopIx < _length && _nextPopIx >= 0);
+            return NULL;
+        }
+
+        // not in this position, check next (and forget this position)
+        _nextPopIx = (_nextPopIx + 1) % _length;
+    }
+
+    // could not find matching timestamp in list
+    assert(_nextPopIx < _length && _nextPopIx >= 0);
+    return NULL;
+}
+
+// Check if no timers are currently running
+bool
+VCMTimestampMap::IsEmpty() const
+{
+    return (_nextAddIx == _nextPopIx);
+}
+
+}

modules/video_coding/main/source/timestamp_map.h

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+struct VCMTimestampDataTuple
+{
+    WebRtc_UWord32    timestamp;
+    void*             data;
+};
+
+class VCMTimestampMap
+{
+public:
+    // Constructor. Optional parameter specifies maximum number of
+    // timestamps in map.
+    VCMTimestampMap(const WebRtc_Word32 length = 10);
+
+    // Destructor.
+    ~VCMTimestampMap();
+
+    // Empty the map
+    void Reset();
+
+    WebRtc_Word32 Add(WebRtc_UWord32 timestamp, void*  data);
+    void* Pop(WebRtc_UWord32 timestamp);
+
+private:
+    bool IsEmpty() const;
+
+    VCMTimestampDataTuple* _map;
+    WebRtc_Word32                   _nextAddIx;
+    WebRtc_Word32                   _nextPopIx;
+    WebRtc_Word32                   _length;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_

modules/video_coding/main/source/timing.cc

@@ -0,0 +1,333 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "trace.h"
+#include "internal_defines.h"
+#include "jitter_buffer_common.h"
+#include "timing.h"
+#include "timestamp_extrapolator.h"
+
+namespace webrtc {
+
+VCMTiming::VCMTiming(WebRtc_Word32 vcmId, WebRtc_Word32 timingId, VCMTiming* masterTiming)
+:
+_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_vcmId(vcmId),
+_timingId(timingId),
+_master(false),
+_tsExtrapolator(),
+_codecTimer(),
+_renderDelayMs(kDefaultRenderDelayMs),
+_minTotalDelayMs(0),
+_requiredDelayMs(0),
+_currentDelayMs(0),
+_prevFrameTimestamp(0)
+{
+    if (masterTiming == NULL)
+    {
+        _master = true;
+        _tsExtrapolator = new VCMTimestampExtrapolator(vcmId, timingId);
+    }
+    else
+    {
+        _tsExtrapolator = masterTiming->_tsExtrapolator;
+    }
+}
+
+VCMTiming::~VCMTiming()
+{
+    if (_master)
+    {
+        delete _tsExtrapolator;
+    }
+    delete &_critSect;
+}
+
+void
+VCMTiming::Reset(WebRtc_Word64 nowMs /* = -1 */)
+{
+    CriticalSectionScoped cs(_critSect);
+    if (nowMs > -1)
+    {
+        _tsExtrapolator->Reset(nowMs);
+    }
+    else
+    {
+        _tsExtrapolator->Reset();
+    }
+    _codecTimer.Reset();
+    _renderDelayMs = kDefaultRenderDelayMs;
+    _minTotalDelayMs = 0;
+    _requiredDelayMs = 0;
+    _currentDelayMs = 0;
+    _prevFrameTimestamp = 0;
+}
+
+void VCMTiming::ResetDecodeTime()
+{
+    _codecTimer.Reset();
+}
+
+void
+VCMTiming::SetRenderDelay(WebRtc_UWord32 renderDelayMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    _renderDelayMs = renderDelayMs;
+}
+
+void
+VCMTiming::SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    _minTotalDelayMs = minTotalDelayMs;
+}
+
+void
+VCMTiming::SetRequiredDelay(WebRtc_UWord32 requiredDelayMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    if (requiredDelayMs != _requiredDelayMs)
+    {
+        if (_master)
+        {
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+                    "Desired jitter buffer level: %u ms", requiredDelayMs);
+        }
+        _requiredDelayMs = requiredDelayMs;
+    }
+}
+
+void VCMTiming::UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp)
+{
+    CriticalSectionScoped cs(_critSect);
+    WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
+
+    // Make sure we try to sync with audio
+    if (targetDelayMs < _minTotalDelayMs)
+    {
+        targetDelayMs = _minTotalDelayMs;
+    }
+
+    if (_currentDelayMs == 0)
+    {
+        // Not initialized, set current delay to target.
+        _currentDelayMs = targetDelayMs;
+    }
+    else if (targetDelayMs != _currentDelayMs)
+    {
+        WebRtc_Word64 delayDiffMs = static_cast<WebRtc_Word64>(targetDelayMs) -
+                                    _currentDelayMs;
+        // Never change the delay with more than 100 ms every second. If we're changing the
+        // delay in too large steps we will get noticable freezes. By limiting the change we
+        // can increase the delay in smaller steps, which will be experienced as the video is
+        // played in slow motion. When lowering the delay the video will be played at a faster
+        // pace.
+        WebRtc_Word64 maxChangeMs = 0;
+        if (frameTimestamp < 0x0000ffff && _prevFrameTimestamp > 0xffff0000)
+        {
+            // wrap
+            maxChangeMs = kDelayMaxChangeMsPerS * (frameTimestamp +
+                         (static_cast<WebRtc_Word64>(1)<<32) - _prevFrameTimestamp) / 90000;
+        }
+        else
+        {
+            maxChangeMs = kDelayMaxChangeMsPerS *
+                          (frameTimestamp - _prevFrameTimestamp) / 90000;
+        }
+        if (maxChangeMs <= 0)
+        {
+            // Any changes less than 1 ms are truncated and
+            // will be postponed. Negative change will be due
+            // to reordering and should be ignored.
+            return;
+        }
+        else if (delayDiffMs < -maxChangeMs)
+        {
+            delayDiffMs = -maxChangeMs;
+        }
+        else if (delayDiffMs > maxChangeMs)
+        {
+            delayDiffMs = maxChangeMs;
+        }
+        _currentDelayMs = _currentDelayMs + static_cast<WebRtc_Word32>(delayDiffMs);
+    }
+    _prevFrameTimestamp = frameTimestamp;
+}
+
+void VCMTiming::UpdateCurrentDelay(WebRtc_Word64 renderTimeMs,
+                                   WebRtc_Word64 actualDecodeTimeMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
+    // Make sure we try to sync with audio
+    if (targetDelayMs < _minTotalDelayMs)
+    {
+        targetDelayMs = _minTotalDelayMs;
+    }
+    WebRtc_Word64 delayedMs = actualDecodeTimeMs -
+                              (renderTimeMs - MaxDecodeTimeMs() - _renderDelayMs);
+    if (delayedMs < 0)
+    {
+        return;
+    }
+    else if (_currentDelayMs + delayedMs <= targetDelayMs)
+    {
+        _currentDelayMs += static_cast<WebRtc_UWord32>(delayedMs);
+    }
+    else
+    {
+        _currentDelayMs = targetDelayMs;
+    }
+}
+
+WebRtc_Word32
+VCMTiming::StopDecodeTimer(WebRtc_UWord32 timeStamp,
+                           WebRtc_Word64 startTimeMs,
+                           WebRtc_Word64 nowMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    const WebRtc_Word32 maxDecTime = MaxDecodeTimeMs();
+    WebRtc_Word32 timeDiffMs = _codecTimer.StopTimer(startTimeMs, nowMs);
+    if (timeDiffMs < 0)
+    {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+            "Codec timer error: %d", timeDiffMs);
+        return timeDiffMs;
+    }
+
+    if (_master)
+    {
+        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+                "Frame decoded: timeStamp=%u decTime=%d maxDecTime=%u, at %u",
+                timeStamp, timeDiffMs, maxDecTime, MaskWord64ToUWord32(nowMs));
+    }
+    return 0;
+}
+
+void
+VCMTiming::IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 nowMs)
+{
+    CriticalSectionScoped cs(_critSect);
+    _tsExtrapolator->Update(nowMs, timeStamp, _master);
+}
+
+WebRtc_Word64
+VCMTiming::RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
+{
+    CriticalSectionScoped cs(_critSect);
+    const WebRtc_Word64 renderTimeMs = RenderTimeMsInternal(frameTimestamp, nowMs);
+    if (renderTimeMs < 0)
+    {
+        return renderTimeMs;
+    }
+    if (_master)
+    {
+        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+            "Render frame %u at %u. Render delay %u, required delay %u,"
+                " max decode time %u, min total delay %u",
+            frameTimestamp, MaskWord64ToUWord32(renderTimeMs), _renderDelayMs,
+            _requiredDelayMs, MaxDecodeTimeMs(),_minTotalDelayMs);
+    }
+    return renderTimeMs;
+}
+
+WebRtc_Word64
+VCMTiming::RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
+{
+    WebRtc_Word64 estimatedCompleteTimeMs =
+            _tsExtrapolator->ExtrapolateLocalTime(frameTimestamp);
+    if (estimatedCompleteTimeMs - nowMs > kMaxVideoDelayMs)
+    {
+        if (_master)
+        {
+            WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+                    "Timestamp arrived 2 seconds early, reset statistics",
+                    frameTimestamp, estimatedCompleteTimeMs);
+        }
+        return -1;
+    }
+    if (_master)
+    {
+        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+                "ExtrapolateLocalTime(%u)=%u ms",
+                frameTimestamp, MaskWord64ToUWord32(estimatedCompleteTimeMs));
+    }
+    if (estimatedCompleteTimeMs == -1)
+    {
+        estimatedCompleteTimeMs = nowMs;
+    }
+
+    return estimatedCompleteTimeMs + _currentDelayMs;
+}
+
+// Must be called from inside a critical section
+WebRtc_Word32
+VCMTiming::MaxDecodeTimeMs(FrameType frameType /*= kVideoFrameDelta*/) const
+{
+    const WebRtc_Word32 decodeTimeMs = _codecTimer.RequiredDecodeTimeMs(frameType);
+
+    if (decodeTimeMs < 0)
+    {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
+            "Negative maximum decode time: %d", decodeTimeMs);
+        return -1;
+    }
+    return decodeTimeMs;
+}
+
+WebRtc_UWord32
+VCMTiming::MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const
+{
+    CriticalSectionScoped cs(_critSect);
+
+    const WebRtc_Word64 maxWaitTimeMs = renderTimeMs - nowMs -
+                                        MaxDecodeTimeMs() - _renderDelayMs;
+
+    if (maxWaitTimeMs < 0)
+    {
+        return 0;
+    }
+    return static_cast<WebRtc_UWord32>(maxWaitTimeMs);
+}
+
+bool
+VCMTiming::EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const
+{
+    CriticalSectionScoped cs(_critSect);
+    WebRtc_Word32 maxDecodeTimeMs = MaxDecodeTimeMs();
+    if (maxDecodeTimeMs < 0)
+    {
+        // Haven't decoded any frames yet, try decoding one to get an estimate
+        // of the decode time.
+        return true;
+    }
+    else if (maxDecodeTimeMs == 0)
+    {
+        // Decode time is less than 1, set to 1 for now since
+        // we don't have any better precision. Count ticks later?
+        maxDecodeTimeMs = 1;
+    }
+    return static_cast<WebRtc_Word32>(availableProcessingTimeMs) - maxDecodeTimeMs > 0;
+}
+
+WebRtc_UWord32
+VCMTiming::TargetVideoDelay() const
+{
+    CriticalSectionScoped cs(_critSect);
+    return TargetDelayInternal();
+}
+
+WebRtc_UWord32
+VCMTiming::TargetDelayInternal() const
+{
+    return _requiredDelayMs + MaxDecodeTimeMs() + _renderDelayMs;
+}
+
+}

modules/video_coding/main/source/timing.h

@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+
+#include "typedefs.h"
+#include "critical_section_wrapper.h"
+#include "codec_timer.h"
+
+namespace webrtc
+{
+
+class VCMTimestampExtrapolator;
+
+class VCMTiming
+{
+public:
+    // The primary timing component should be passed
+    // if this is the dual timing component.
+    VCMTiming(WebRtc_Word32 vcmId = 0,
+              WebRtc_Word32 timingId = 0,
+              VCMTiming* masterTiming = NULL);
+    ~VCMTiming();
+
+    // Resets the timing to the initial state.
+    void Reset(WebRtc_Word64 nowMs = -1);
+    void ResetDecodeTime();
+
+    // The amount of time needed to render an image. Defaults to 10 ms.
+    void SetRenderDelay(WebRtc_UWord32 renderDelayMs);
+
+    // The minimum time the video must be delayed on the receiver to
+    // get the desired jitter buffer level.
+    void SetRequiredDelay(WebRtc_UWord32 requiredDelayMs);
+
+    // Minimum total delay required to sync video with audio.
+    void SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs);
+
+    // Increases or decreases the current delay to get closer to the target delay.
+    // Calculates how long it has been since the previous call to this function,
+    // and increases/decreases the delay in proportion to the time difference.
+    void UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp);
+
+    // Increases or decreases the current delay to get closer to the target delay.
+    // Given the actual decode time in ms and the render time in ms for a frame, this
+    // function calculates how late the frame is and increases the delay accordingly.
+    void UpdateCurrentDelay(WebRtc_Word64 renderTimeMs, WebRtc_Word64 actualDecodeTimeMs);
+
+    // Stops the decoder timer, should be called when the decoder returns a frame
+    // or when the decoded frame callback is called.
+    WebRtc_Word32 StopDecodeTimer(WebRtc_UWord32 timeStamp,
+                                  WebRtc_Word64 startTimeMs,
+                                  WebRtc_Word64 nowMs);
+
+    // Used to report that a frame is passed to decoding. Updates the timestamp filter
+    // which is used to map between timestamps and receiver system time.
+    void IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 lastPacketTimeMs);
+
+    // Returns the receiver system time when the frame with timestamp frameTimestamp
+    // should be rendered, assuming that the system time currently is nowMs.
+    WebRtc_Word64 RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const;
+
+    // Returns the maximum time in ms that we can wait for a frame to become complete
+    // before we must pass it to the decoder.
+    WebRtc_UWord32 MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const;
+
+    // Returns the current target delay which is required delay + decode time + render
+    // delay.
+    WebRtc_UWord32 TargetVideoDelay() const;
+
+    // Calculates whether or not there is enough time to decode a frame given a
+    // certain amount of processing time.
+    bool EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const;
+
+    enum { kDefaultRenderDelayMs = 10 };
+    enum { kDelayMaxChangeMsPerS = 100 };
+
+protected:
+    WebRtc_Word32 MaxDecodeTimeMs(FrameType frameType = kVideoFrameDelta) const;
+    WebRtc_Word64 RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp,
+                                       WebRtc_Word64 nowMs) const;
+    WebRtc_UWord32 TargetDelayInternal() const;
+
+private:
+    CriticalSectionWrapper&          _critSect;
+    WebRtc_Word32                 _vcmId;
+    WebRtc_Word32                 _timingId;
+    bool                          _master;
+    VCMTimestampExtrapolator*     _tsExtrapolator;
+    VCMCodecTimer                 _codecTimer;
+    WebRtc_UWord32                _renderDelayMs;
+    WebRtc_UWord32                _minTotalDelayMs;
+    WebRtc_UWord32                _requiredDelayMs;
+    WebRtc_UWord32                _currentDelayMs;
+    WebRtc_UWord32                _prevFrameTimestamp;
+    WebRtc_Word64                 _startStoragePlaybackMs;
+    WebRtc_Word64                 _firstStoredRenderTimeMs;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_

modules/video_coding/main/source/video_coding.gyp

@@ -0,0 +1,103 @@
+# Copyright (c) 2009 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+  'includes': [
+    '../../../../common_settings.gypi', # Common settings
+  ],
+  'targets': [
+    {
+      'target_name': 'webrtc_video_coding',
+      'type': '<(library)',
+      'dependencies': [
+        '../../codecs/i420/main/source/i420.gyp:webrtc_i420',
+        '../../codecs/vp8/main/source/vp8.gyp:webrtc_vp8',
+        '../../../../common_video/vplib/main/source/vplib.gyp:webrtc_vplib',
+        '../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
+      ],
+      'include_dirs': [
+        '../interface',
+        '../../../interface',
+        '../../codecs/interface',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '../interface',
+          '../../codecs/interface',
+        ],
+      },
+      'sources': [
+        # interfaces
+        '../interface/video_coding.h',
+        '../interface/video_coding_defines.h',
+
+        # headers
+        'codec_database.h',
+        'codec_timer.h',
+        'content_metrics_processing.h',
+        'encoded_frame.h',
+        'er_tables_xor.h',
+        'event.h',
+        'exp_filter.h',
+        'fec_tables_xor.h',
+        'frame_buffer.h',
+        'frame_dropper.h',
+        'frame_list.h',
+        'generic_decoder.h',
+        'generic_encoder.h',
+        'inter_frame_delay.h',
+        'internal_defines.h',
+        'jitter_buffer_common.h',
+        'jitter_buffer.h',
+        'jitter_estimator.h',
+        'media_opt_util.h',
+        'media_optimization.h',
+        'nack_fec_tables.h',
+        'packet.h',
+        'qm_select_data.h',
+        'qm_select.h',
+        'receiver.h',
+        'rtt_filter.h',
+        'session_info.h',
+        'tick_time.h',
+        'timestamp_extrapolator.h',
+        'timestamp_map.h',
+        'timing.h',
+        'video_coding_impl.h',
+
+        # sources
+        'codec_database.cc',
+        'codec_timer.cc',
+        'content_metrics_processing.cc',
+        'encoded_frame.cc',
+        'exp_filter.cc',
+        'frame_buffer.cc',
+        'frame_dropper.cc',
+        'frame_list.cc',
+        'generic_decoder.cc',
+        'generic_encoder.cc',
+        'inter_frame_delay.cc',
+        'jitter_buffer.cc',
+        'jitter_estimator.cc',
+        'media_opt_util.cc',
+        'media_optimization.cc',
+        'packet.cc',
+        'qm_select.cc',
+        'receiver.cc',
+        'rtt_filter.cc',
+        'session_info.cc',
+        'timestamp_extrapolator.cc',
+        'timestamp_map.cc',
+        'timing.cc',
+        'video_coding_impl.cc',
+      ], # source
+    },
+  ],
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2: