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// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#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_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"
namespace content {
WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
WebRtcLocalAudioTrackAdapter* adapter,
const scoped_refptr<WebRtcAudioCapturer>& capturer,
WebAudioCapturerSource* webaudio_source)
: MediaStreamTrackExtraData(adapter, true),
adapter_(adapter),
capturer_(capturer),
webaudio_source_(webaudio_source) {
DCHECK(capturer.get() || webaudio_source);
adapter_->Initialize(this);
if (!webaudio_source_) {
source_provider_.reset(new WebRtcLocalAudioSourceProvider());
AddSink(source_provider_.get());
}
DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
}
WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
// Users might not call Stop() on the track.
Stop();
}
void WebRtcLocalAudioTrack::Capture(const int16* audio_data,
base::TimeDelta delay,
int volume,
bool key_pressed,
bool need_audio_processing) {
DCHECK(capture_thread_checker_.CalledOnValidThread());
scoped_refptr<WebRtcAudioCapturer> capturer;
SinkList::ItemList sinks;
SinkList::ItemList sinks_to_notify_format;
{
base::AutoLock auto_lock(lock_);
capturer = capturer_;
sinks = sinks_.Items();
sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
}
// Notify the tracks on when the format changes. This will do nothing if
// |sinks_to_notify_format| is empty.
for (SinkList::ItemList::const_iterator it = sinks_to_notify_format.begin();
it != sinks_to_notify_format.end(); ++it) {
(*it)->OnSetFormat(audio_parameters_);
}
// Feed the data to the sinks.
// TODO(jiayl): we should not pass the real audio data down if the track is
// 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) {
int new_volume = (*it)->OnData(audio_data,
audio_parameters_.sample_rate(),
audio_parameters_.channels(),
audio_parameters_.frames_per_buffer(),
voe_channels,
delay.InMilliseconds(),
volume,
need_audio_processing,
key_pressed);
if (new_volume != 0 && capturer.get() && !webaudio_source_) {
// Feed the new volume to WebRtc while changing the volume on the
// browser.
capturer->SetVolume(new_volume);
}
}
}
void WebRtcLocalAudioTrack::OnSetFormat(
const media::AudioParameters& params) {
DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
// If the source is restarted, we might have changed to another capture
// thread.
capture_thread_checker_.DetachFromThread();
DCHECK(capture_thread_checker_.CalledOnValidThread());
audio_parameters_ = params;
base::AutoLock auto_lock(lock_);
// Remember to notify all sinks of the new format.
sinks_.TagAll();
}
void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
base::AutoLock auto_lock(lock_);
// Verify that |sink| is not already added to the list.
DCHECK(!sinks_.Contains(
MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
// Create (and add to the list) a new MediaStreamAudioTrackSink
// which owns the |sink| and delagates all calls to the
// MediaStreamAudioSink interface. It will be tagged in the list, so
// we remember to call OnSetFormat() on the new sink.
scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
new MediaStreamAudioSinkOwner(sink));
sinks_.AddAndTag(sink_owner);
}
void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
base::AutoLock auto_lock(lock_);
scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove(
MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
// Clear the delegate to ensure that no more capture callbacks will
// be sent to this sink. Also avoids a possible crash which can happen
// if this method is called while capturing is active.
if (removed_item.get())
removed_item->Reset();
}
void WebRtcLocalAudioTrack::AddSink(PeerConnectionAudioSink* sink) {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
base::AutoLock auto_lock(lock_);
// Verify that |sink| is not already added to the list.
DCHECK(!sinks_.Contains(
MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink)));
// Create (and add to the list) a new MediaStreamAudioTrackSink
// which owns the |sink| and delagates all calls to the
// MediaStreamAudioSink interface. It will be tagged in the list, so
// we remember to call OnSetFormat() on the new sink.
scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
new PeerConnectionAudioSinkOwner(sink));
sinks_.AddAndTag(sink_owner);
}
void WebRtcLocalAudioTrack::RemoveSink(PeerConnectionAudioSink* sink) {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
base::AutoLock auto_lock(lock_);
scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove(
MediaStreamAudioTrackSink::WrapsPeerConnectionSink(sink));
// Clear the delegate to ensure that no more capture callbacks will
// be sent to this sink. Also avoids a possible crash which can happen
// if this method is called while capturing is active.
if (removed_item.get())
removed_item->Reset();
}
void WebRtcLocalAudioTrack::Start() {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
if (webaudio_source_.get()) {
// If the track is hooking up with WebAudio, do NOT add the track to the
// capturer as its sink otherwise two streams in different clock will be
// pushed through the same track.
webaudio_source_->Start(this, capturer_.get());
} else if (capturer_.get()) {
capturer_->AddTrack(this);
}
SinkList::ItemList sinks;
{
base::AutoLock auto_lock(lock_);
sinks = sinks_.Items();
}
for (SinkList::ItemList::const_iterator it = sinks.begin();
it != sinks.end();
++it) {
(*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive);
}
}
void WebRtcLocalAudioTrack::Stop() {
DCHECK(main_render_thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
if (!capturer_.get() && !webaudio_source_.get())
return;
if (webaudio_source_.get()) {
// Called Stop() on the |webaudio_source_| explicitly so that
// |webaudio_source_| won't push more data to the track anymore.
// Also note that the track is not registered as a sink to the |capturer_|
// in such case and no need to call RemoveTrack().
webaudio_source_->Stop();
} else {
// It is necessary to call RemoveTrack on the |capturer_| to avoid getting
// audio callback after Stop().
capturer_->RemoveTrack(this);
}
// Protect the pointers using the lock when accessing |sinks_| and
// setting the |capturer_| to NULL.
SinkList::ItemList sinks;
{
base::AutoLock auto_lock(lock_);
sinks = sinks_.Items();
sinks_.Clear();
webaudio_source_ = NULL;
capturer_ = NULL;
}
for (SinkList::ItemList::const_iterator it = sinks.begin();
it != sinks.end();
++it){
(*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
(*it)->Reset();
}
}
} // namespace content
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