#include "content/renderer/media/webrtc_local_audio_track.h"
#include "content/public/renderer/media_stream_audio_sink.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
+#include "content/renderer/media/media_stream_audio_processor.h"
#include "content/renderer/media/media_stream_audio_sink_owner.h"
#include "content/renderer/media/media_stream_audio_track_sink.h"
#include "content/renderer/media/peer_connection_audio_sink_owner.h"
#include "content/renderer/media/webaudio_capturer_source.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_local_audio_source_provider.h"
-#include "media/base/audio_fifo.h"
-#include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
namespace content {
-static const char kAudioTrackKind[] = "audio";
-
-scoped_refptr<WebRtcLocalAudioTrack> WebRtcLocalAudioTrack::Create(
- const std::string& id,
- const scoped_refptr<WebRtcAudioCapturer>& capturer,
- WebAudioCapturerSource* webaudio_source,
- webrtc::AudioSourceInterface* track_source) {
- talk_base::RefCountedObject<WebRtcLocalAudioTrack>* track =
- new talk_base::RefCountedObject<WebRtcLocalAudioTrack>(
- id, capturer, webaudio_source, track_source);
- return track;
-}
-
WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
- const std::string& label,
+ WebRtcLocalAudioTrackAdapter* adapter,
const scoped_refptr<WebRtcAudioCapturer>& capturer,
- WebAudioCapturerSource* webaudio_source,
- webrtc::AudioSourceInterface* track_source)
- : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
+ WebAudioCapturerSource* webaudio_source)
+ : MediaStreamTrack(adapter, true),
+ adapter_(adapter),
capturer_(capturer),
- webaudio_source_(webaudio_source),
- track_source_(track_source) {
+ webaudio_source_(webaudio_source) {
DCHECK(capturer.get() || webaudio_source);
- if (!webaudio_source_) {
- source_provider_.reset(new WebRtcLocalAudioSourceProvider());
- AddSink(source_provider_.get());
- }
+
+ adapter_->Initialize(this);
+
DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
}
base::TimeDelta delay,
int volume,
bool key_pressed,
- bool need_audio_processing) {
+ bool need_audio_processing,
+ bool force_report_nonzero_energy) {
DCHECK(capture_thread_checker_.CalledOnValidThread());
+
+ // Calculate the signal level regardless if the track is disabled or enabled.
+ int signal_level = level_calculator_->Calculate(
+ audio_data, audio_parameters_.channels(),
+ audio_parameters_.frames_per_buffer(), force_report_nonzero_energy);
+ adapter_->SetSignalLevel(signal_level);
+
scoped_refptr<WebRtcAudioCapturer> capturer;
- std::vector<int> voe_channels;
SinkList::ItemList sinks;
SinkList::ItemList sinks_to_notify_format;
{
base::AutoLock auto_lock(lock_);
capturer = capturer_;
- voe_channels = voe_channels_;
sinks = sinks_.Items();
sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
}
// disabled. This is currently done so to feed input to WebRTC typing
// detection and should be changed when audio processing is moved from
// WebRTC to the track.
+ std::vector<int> voe_channels = adapter_->VoeChannels();
for (SinkList::ItemList::const_iterator it = sinks.begin();
it != sinks.end();
++it) {
volume,
need_audio_processing,
key_pressed);
- if (new_volume != 0 && capturer.get() && !webaudio_source_) {
+ if (new_volume != 0 && capturer.get() && !webaudio_source_.get()) {
// Feed the new volume to WebRtc while changing the volume on the
// browser.
capturer->SetVolume(new_volume);
DCHECK(capture_thread_checker_.CalledOnValidThread());
audio_parameters_ = params;
+ level_calculator_.reset(new MediaStreamAudioLevelCalculator());
base::AutoLock auto_lock(lock_);
// Remember to notify all sinks of the new format.
sinks_.TagAll();
}
-void WebRtcLocalAudioTrack::AddChannel(int channel_id) {
- DVLOG(1) << "WebRtcLocalAudioTrack::AddChannel(channel_id="
- << channel_id << ")";
- base::AutoLock auto_lock(lock_);
- if (std::find(voe_channels_.begin(), voe_channels_.end(), channel_id) !=
- voe_channels_.end()) {
- // We need to handle the case when the same channel is connected to the
- // track more than once.
- return;
- }
-
- voe_channels_.push_back(channel_id);
-}
-
-void WebRtcLocalAudioTrack::RemoveChannel(int channel_id) {
- DVLOG(1) << "WebRtcLocalAudioTrack::RemoveChannel(channel_id="
- << channel_id << ")";
- base::AutoLock auto_lock(lock_);
- std::vector<int>::iterator iter =
- std::find(voe_channels_.begin(), voe_channels_.end(), channel_id);
- DCHECK(iter != voe_channels_.end());
- voe_channels_.erase(iter);
-}
-
-// webrtc::AudioTrackInterface implementation.
-webrtc::AudioSourceInterface* WebRtcLocalAudioTrack::GetSource() const {
- return track_source_;
-}
-
-cricket::AudioRenderer* WebRtcLocalAudioTrack::GetRenderer() {
- return this;
-}
-
-std::string WebRtcLocalAudioTrack::kind() const {
- return kAudioTrackKind;
+void WebRtcLocalAudioTrack::SetAudioProcessor(
+ const scoped_refptr<MediaStreamAudioProcessor>& processor) {
+ // if the |processor| does not have audio processing, which can happen if
+ // kDisableAudioTrackProcessing is set set or all the constraints in
+ // the |processor| are turned off. In such case, we pass NULL to the
+ // adapter to indicate that no stats can be gotten from the processor.
+ adapter_->SetAudioProcessor(processor->has_audio_processing() ?
+ processor : NULL);
}
void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
// we remember to call OnSetFormat() on the new sink.
scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
new MediaStreamAudioSinkOwner(sink));
- sinks_.AddAndTag(sink_owner);
+ sinks_.AddAndTag(sink_owner.get());
}
void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
// we remember to call OnSetFormat() on the new sink.
scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
new PeerConnectionAudioSinkOwner(sink));
- sinks_.AddAndTag(sink_owner);
+ sinks_.AddAndTag(sink_owner.get());
}
void WebRtcLocalAudioTrack::RemoveSink(PeerConnectionAudioSink* sink) {
projects / platform / framework / web / crosswalk.git / blobdiff