content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc

   1 // Copyright 2014 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "base/logging.h"
   6 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
   7 #include "media/base/audio_bus.h"
   8 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
   9
  10 namespace content {
  11
  12 WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
  13     webrtc::AudioTrackSinkInterface* sink)
  14     : sink_(sink) {
  15   DCHECK(sink);
  16 }
  17
  18 WebRtcAudioSinkAdapter::~WebRtcAudioSinkAdapter() {
  19 }
  20
  21 bool WebRtcAudioSinkAdapter::IsEqual(
  22     const webrtc::AudioTrackSinkInterface* other) const {
  23   return (other == sink_);
  24 }
  25
  26 void WebRtcAudioSinkAdapter::OnData(const media::AudioBus& audio_bus,
  27                                     base::TimeTicks estimated_capture_time) {
  28   DCHECK_EQ(audio_bus.frames(), params_.frames_per_buffer());
  29   DCHECK_EQ(audio_bus.channels(), params_.channels());
  30   // TODO(henrika): Remove this conversion once the interface in libjingle
  31   // supports float vectors.
  32   audio_bus.ToInterleaved(audio_bus.frames(),
  33                           sizeof(interleaved_data_[0]),
  34                           interleaved_data_.get());
  35   sink_->OnData(interleaved_data_.get(),
  36                 16,
  37                 params_.sample_rate(),
  38                 audio_bus.channels(),
  39                 audio_bus.frames());
  40 }
  41
  42 void WebRtcAudioSinkAdapter::OnSetFormat(
  43     const media::AudioParameters& params) {
  44   DCHECK(params.IsValid());
  45   params_ = params;
  46   const int num_pcm16_data_elements =
  47       params_.frames_per_buffer() * params_.channels();
  48   interleaved_data_.reset(new int16[num_pcm16_data_elements]);
  49 }
  50
  51 }  // namespace content