/*
|
* libjingle
|
* Copyright 2004--2011 Google Inc.
|
*
|
* Redistribution and use in source and binary forms, with or without
|
* modification, are permitted provided that the following conditions are met:
|
*
|
* 1. Redistributions of source code must retain the above copyright notice,
|
* this list of conditions and the following disclaimer.
|
* 2. Redistributions in binary form must reproduce the above copyright notice,
|
* this list of conditions and the following disclaimer in the documentation
|
* and/or other materials provided with the distribution.
|
* 3. The name of the author may not be used to endorse or promote products
|
* derived from this software without specific prior written permission.
|
*
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
|
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
|
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
|
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
|
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
|
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
|
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
*/
|
|
#include "talk/app/webrtc/audiotrack.h"
|
|
#include "webrtc/base/checks.h"
|
|
using rtc::scoped_refptr;
|
|
namespace webrtc {
|
|
const char MediaStreamTrackInterface::kAudioKind[] = "audio";
|
|
// static
|
scoped_refptr<AudioTrack> AudioTrack::Create(
|
const std::string& id,
|
const scoped_refptr<AudioSourceInterface>& source) {
|
return new rtc::RefCountedObject<AudioTrack>(id, source);
|
}
|
|
AudioTrack::AudioTrack(const std::string& label,
|
const scoped_refptr<AudioSourceInterface>& source)
|
: MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
|
if (audio_source_) {
|
audio_source_->RegisterObserver(this);
|
OnChanged();
|
}
|
}
|
|
AudioTrack::~AudioTrack() {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
set_state(MediaStreamTrackInterface::kEnded);
|
if (audio_source_)
|
audio_source_->UnregisterObserver(this);
|
}
|
|
std::string AudioTrack::kind() const {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
return kAudioKind;
|
}
|
|
AudioSourceInterface* AudioTrack::GetSource() const {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
return audio_source_.get();
|
}
|
|
void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
if (audio_source_)
|
audio_source_->AddSink(sink);
|
}
|
|
void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
if (audio_source_)
|
audio_source_->RemoveSink(sink);
|
}
|
|
void AudioTrack::OnChanged() {
|
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
if (state() == kFailed)
|
return; // We can't recover from this state (do we ever set it?).
|
|
TrackState new_state = kInitializing;
|
|
// |audio_source_| must be non-null if we ever get here.
|
switch (audio_source_->state()) {
|
case MediaSourceInterface::kLive:
|
case MediaSourceInterface::kMuted:
|
new_state = kLive;
|
break;
|
case MediaSourceInterface::kEnded:
|
new_state = kEnded;
|
break;
|
case MediaSourceInterface::kInitializing:
|
default:
|
// use kInitializing.
|
break;
|
}
|
|
set_state(new_state);
|
}
|
|
} // namespace webrtc
|
