Mainly hlundin's patch.

Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2012 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 "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+
+bool InputAudioFile::Read(size_t samples, int16_t* destination) {
+  if (!fp_) {
+    return false;
+  }
+  size_t bytes_read = fread(destination, sizeof(int16_t), samples, fp_);
+  if (bytes_read < samples) {
+    // Rewind and read the missing sampels.
+    rewind(fp_);
+    size_t missing_samples = samples - bytes_read;
+    if (fread(destination, sizeof(int16_t), missing_samples, fp_) <
+        missing_samples) {
+      // Could not read enough even after rewinding the file.
+      return false;
+    }
+  }
+  return true;
+}
+
+void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples,
+                                          size_t channels,
+                                          int16_t* destination) {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t j = 0; j < channels; ++j) {
+      destination[i * channels + j] = source[i];
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/input_audio_file.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2012 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_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+
+#include <cstdio>
+#include <string>
+
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file.
+class InputAudioFile {
+ public:
+  explicit InputAudioFile(const std::string file_name) {
+    fp_ = fopen(file_name.c_str(), "rb");
+  }
+
+  virtual ~InputAudioFile() {
+    fclose(fp_);
+  }
+
+  // Reads |samples| elements from source file to |destination|. Returns true
+  // if the read was successful, otherwise false. If the file end is reached,
+  // the file is rewound and reading continues from the beginning.
+  // The output |destination| must have the capacity to hold |samples| elements.
+  bool Read(size_t samples, int16_t* destination);
+
+  // Creates a multi-channel signal from a mono signal. Each sample is repeated
+  // |channels| times to create an interleaved multi-channel signal where all
+  // channels are identical. The output |destination| must have the capacity to
+  // hold samples * channels elements.
+  static void DuplicateInterleaved(const int16_t* source, size_t samples,
+                                   size_t channels, int16_t* destination);
+
+ private:
+  FILE* fp_;
+  DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_

webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2012 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 "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+
+namespace webrtc {
+namespace test {
+
+uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                    size_t payload_length_samples,
+                                    WebRtcRTPHeader* rtp_header) {
+  assert(rtp_header);
+  if (!rtp_header) {
+    return 0;
+  }
+  rtp_header->header.sequenceNumber = seq_number_++;
+  rtp_header->header.timestamp = timestamp_;
+  timestamp_ += payload_length_samples;
+  rtp_header->header.payloadType = payload_type;
+  rtp_header->header.markerBit = false;
+  rtp_header->header.ssrc = ssrc_;
+  rtp_header->header.numCSRCs = 0;
+  rtp_header->frameType = kAudioFrameSpeech;
+
+  uint32_t this_send_time = next_send_time_ms_;
+  assert(samples_per_ms_ > 0);
+  next_send_time_ms_ += payload_length_samples / samples_per_ms_;
+  return this_send_time;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/rtp_generator.h

@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2012 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_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for generating RTP headers.
+class RtpGenerator {
+ public:
+  RtpGenerator(int samples_per_ms,
+               uint16_t start_seq_number = 0,
+               uint32_t start_timestamp = 0,
+               uint32_t start_send_time_ms = 0,
+               uint32_t ssrc = 0x12345678)
+      : seq_number_(start_seq_number),
+        timestamp_(start_timestamp),
+        next_send_time_ms_(start_send_time_ms),
+        ssrc_(ssrc),
+        samples_per_ms_(samples_per_ms) {
+  }
+
+  // Writes the next RTP header to |rtp_header|, wich will be of type
+  // |payload_type|. Returns the send time for this packet (in ms). The value of
+  // |payload_length_samples| determines the send time for the next packet.
+  uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples,
+                        WebRtcRTPHeader* rtp_header);
+
+ private:
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  uint32_t next_send_time_ms_;
+  const uint32_t ssrc_;
+  const int samples_per_ms_;
+  DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_