| Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| index 56729197f1b18a1db5ba95a4d77ee8214fed3324..b1bde99a258b9555502a34a02c61353d58937be6 100644
|
| --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| @@ -5,8 +5,10 @@
|
| #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
|
|
|
| #include "base/logging.h"
|
| +#include "content/renderer/media/media_stream_audio_processor.h"
|
| #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
|
| #include "content/renderer/media/webrtc_local_audio_track.h"
|
| +#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
|
|
| namespace content {
|
|
|
| @@ -27,7 +29,8 @@ WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
|
| webrtc::AudioSourceInterface* track_source)
|
| : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
|
| owner_(NULL),
|
| - track_source_(track_source) {
|
| + track_source_(track_source),
|
| + signal_level_(0) {
|
| }
|
|
|
| WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
|
| @@ -39,6 +42,12 @@ void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
|
| owner_ = owner;
|
| }
|
|
|
| +void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
|
| + const scoped_refptr<MediaStreamAudioProcessor>& processor) {
|
| + base::AutoLock auto_lock(lock_);
|
| + audio_processor_ = processor;
|
| +}
|
| +
|
| std::string WebRtcLocalAudioTrackAdapter::kind() const {
|
| return kAudioTrackKind;
|
| }
|
| @@ -75,13 +84,26 @@ void WebRtcLocalAudioTrackAdapter::RemoveSink(
|
| }
|
| }
|
|
|
| +bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
|
| + base::AutoLock auto_lock(lock_);
|
| + *level = signal_level_;
|
| + return true;
|
| +}
|
| +
|
| +talk_base::scoped_refptr<webrtc::AudioProcessorInterface>
|
| +WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
|
| + base::AutoLock auto_lock(lock_);
|
| + return audio_processor_.get();
|
| +}
|
| +
|
| std::vector<int> WebRtcLocalAudioTrackAdapter::VoeChannels() const {
|
| base::AutoLock auto_lock(lock_);
|
| return voe_channels_;
|
| }
|
|
|
| void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int signal_level) {
|
| - // TODO(xians): Implements this.
|
| + base::AutoLock auto_lock(lock_);
|
| + signal_level_ = signal_level;
|
| }
|
|
|
| void WebRtcLocalAudioTrackAdapter::AddChannel(int channel_id) {
|
|
|