Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_device/include/audio_device_defines.h

@@ -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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+
+#include <stddef.h>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceObserver
+// ----------------------------------------------------------------------------
+
+class AudioDeviceObserver {
+ public:
+  enum ErrorCode { kRecordingError = 0, kPlayoutError = 1 };
+  enum WarningCode { kRecordingWarning = 0, kPlayoutWarning = 1 };
+
+  virtual void OnErrorIsReported(const ErrorCode error) = 0;
+  virtual void OnWarningIsReported(const WarningCode warning) = 0;
+
+ protected:
+  virtual ~AudioDeviceObserver() {}
+};
+
+// ----------------------------------------------------------------------------
+//  AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport {
+ public:
+  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                          const size_t nSamples,
+                                          const size_t nBytesPerSample,
+                                          const size_t nChannels,
+                                          const uint32_t samplesPerSec,
+                                          const uint32_t totalDelayMS,
+                                          const int32_t clockDrift,
+                                          const uint32_t currentMicLevel,
+                                          const bool keyPressed,
+                                          uint32_t& newMicLevel) = 0;
+
+  virtual int32_t NeedMorePlayData(const size_t nSamples,
+                                   const size_t nBytesPerSample,
+                                   const size_t nChannels,
+                                   const uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   size_t& nSamplesOut,
+                                   int64_t* elapsed_time_ms,
+                                   int64_t* ntp_time_ms) = 0;
+
+  // Method to push the captured audio data to the specific VoE channel.
+  // The data will not undergo audio processing.
+  // |voe_channel| is the id of the VoE channel which is the sink to the
+  // capture data.
+  virtual void PushCaptureData(int voe_channel,
+                               const void* audio_data,
+                               int bits_per_sample,
+                               int sample_rate,
+                               size_t number_of_channels,
+                               size_t number_of_frames) = 0;
+
+  // Method to pull mixed render audio data from all active VoE channels.
+  // The data will not be passed as reference for audio processing internally.
+  // TODO(xians): Support getting the unmixed render data from specific VoE
+  // channel.
+  virtual void PullRenderData(int bits_per_sample,
+                              int sample_rate,
+                              size_t number_of_channels,
+                              size_t number_of_frames,
+                              void* audio_data,
+                              int64_t* elapsed_time_ms,
+                              int64_t* ntp_time_ms) = 0;
+
+ protected:
+  virtual ~AudioTransport() {}
+};
+
+// Helper class for storage of fundamental audio parameters such as sample rate,
+// number of channels, native buffer size etc.
+// Note that one audio frame can contain more than one channel sample and each
+// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
+// stereo contains 2 * (16/8) = 4 bytes of data.
+class AudioParameters {
+ public:
+  // This implementation does only support 16-bit PCM samples.
+  static const size_t kBitsPerSample = 16;
+  AudioParameters()
+      : sample_rate_(0),
+        channels_(0),
+        frames_per_buffer_(0),
+        frames_per_10ms_buffer_(0) {}
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
+      : sample_rate_(sample_rate),
+        channels_(channels),
+        frames_per_buffer_(frames_per_buffer),
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
+    sample_rate_ = sample_rate;
+    channels_ = channels;
+    frames_per_buffer_ = frames_per_buffer;
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
+  }
+  size_t bits_per_sample() const { return kBitsPerSample; }
+  void reset(int sample_rate, size_t channels, double ms_per_buffer) {
+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
+  }
+  void reset(int sample_rate, size_t channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));
+  }
+  int sample_rate() const { return sample_rate_; }
+  size_t channels() const { return channels_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
+    return frames_per_buffer_ * GetBytesPerFrame();
+  }
+  // The WebRTC audio device buffer (ADB) only requires that the sample rate
+  // and number of channels are configured. Hence, to be "valid", only these
+  // two attributes must be set.
+  bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
+  size_t GetBytesPer10msBuffer() const {
+    return frames_per_10ms_buffer_ * GetBytesPerFrame();
+  }
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
+  }
+
+ private:
+  int sample_rate_;
+  size_t channels_;
+  size_t frames_per_buffer_;
+  size_t frames_per_10ms_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_