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 41
