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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/renderer/media/webrtc_audio_capturer.h"
#include "content/renderer/media/webrtc_audio_capturer_sink_owner.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,
webrtc::AudioSourceInterface* track_source) {
talk_base::RefCountedObject<WebRtcLocalAudioTrack>* track =
new talk_base::RefCountedObject<WebRtcLocalAudioTrack>(
id, capturer, track_source);
return track;
}
WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
const std::string& label,
const scoped_refptr<WebRtcAudioCapturer>& capturer,
webrtc::AudioSourceInterface* track_source)
: webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
capturer_(capturer),
track_source_(track_source),
need_audio_processing_(!capturer->device_id().empty()) {
// The capturer with a valid device id is using microphone as source,
// and APM (AudioProcessingModule) is turned on only for microphone data.
DCHECK(capturer.get());
DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
}
WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
// Users might not call Stop() on the track.
Stop();
}
void WebRtcLocalAudioTrack::CaptureData(const int16* audio_data,
int number_of_channels,
int number_of_frames,
int audio_delay_milliseconds,
int volume,
bool key_pressed) {
scoped_refptr<WebRtcAudioCapturer> capturer;
std::vector<int> voe_channels;
int sample_rate = 0;
SinkList sinks;
{
base::AutoLock auto_lock(lock_);
// When the track is diabled, we simply return here.
// TODO(xians): Figure out if we should feed zero to sinks instead, in
// order to inject VAD data in such case.
if (!enabled())
return;
capturer = capturer_;
voe_channels = voe_channels_;
sample_rate = params_.sample_rate(),
sinks = sinks_;
}
// Feed the data to the sinks.
for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it) {
int new_volume = (*it)->CaptureData(voe_channels,
audio_data,
sample_rate,
number_of_channels,
number_of_frames,
audio_delay_milliseconds,
volume,
need_audio_processing_,
key_pressed);
if (new_volume != 0 && capturer.get())
capturer->SetVolume(new_volume);
}
}
void WebRtcLocalAudioTrack::SetCaptureFormat(
const media::AudioParameters& params) {
base::AutoLock auto_lock(lock_);
params_ = params;
// Update all the existing sinks with the new format.
for (SinkList::const_iterator it = sinks_.begin();
it != sinks_.end(); ++it)
(*it)->SetCaptureFormat(params);
}
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::AddSink(WebRtcAudioCapturerSink* sink) {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
base::AutoLock auto_lock(lock_);
sink->SetCaptureFormat(params_);
// Verify that |sink| is not already added to the list.
DCHECK(std::find_if(
sinks_.begin(), sinks_.end(),
WebRtcAudioCapturerSinkOwner::WrapsSink(sink)) == sinks_.end());
// Create (and add to the list) a new WebRtcAudioCapturerSinkOwner which owns
// the |sink| and delagates all calls to the WebRtcAudioCapturerSink
// interface.
sinks_.push_back(new WebRtcAudioCapturerSinkOwner(sink));
}
void WebRtcLocalAudioTrack::RemoveSink(
WebRtcAudioCapturerSink* sink) {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
base::AutoLock auto_lock(lock_);
// Get iterator to the first element for which WrapsSink(sink) returns true.
SinkList::iterator it = std::find_if(
sinks_.begin(), sinks_.end(),
WebRtcAudioCapturerSinkOwner::WrapsSink(sink));
if (it != sinks_.end()) {
// 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.
(*it)->Reset();
sinks_.erase(it);
}
}
void WebRtcLocalAudioTrack::Start() {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
if (capturer_.get())
capturer_->AddTrack(this);
}
void WebRtcLocalAudioTrack::Stop() {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
if (!capturer_.get())
return;
capturer_->RemoveTrack(this);
// Protect the pointers using the lock when accessing |sinks_| and
// setting the |capturer_| to NULL.
SinkList sinks;
{
base::AutoLock auto_lock(lock_);
sinks = sinks_;
capturer_ = NULL;
}
for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it)
(*it)->Reset();
}
} // namespace content
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