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 "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
   8
   9 namespace content {
  10
  11 WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
  12     webrtc::AudioTrackSinkInterface* sink)
  13     : sink_(sink) {
  14   DCHECK(sink);
  15 }
  16
  17 WebRtcAudioSinkAdapter::~WebRtcAudioSinkAdapter() {
  18 }
  19
  20 bool WebRtcAudioSinkAdapter::IsEqual(
  21     const webrtc::AudioTrackSinkInterface* other) const {
  22   return (other == sink_);
  23 }
  24
  25 void WebRtcAudioSinkAdapter::OnData(const int16* audio_data,
  26                                     int sample_rate,
  27                                     int number_of_channels,
  28                                     int number_of_frames) {
  29   sink_->OnData(audio_data, 16, sample_rate, number_of_channels,
  30                 number_of_frames);
  31 }
  32
  33 void WebRtcAudioSinkAdapter::OnSetFormat(
  34     const media::AudioParameters& params) {
  35   // No need to forward the OnSetFormat() callback to
  36   // webrtc::AudioTrackSinkInterface sink since the sink will handle the
  37   // format change in OnData().
  38 }
  39
  40 }  // namespace content