b24317bfda
In addition to the code moved from talk/app/webrtc there were some files in webrtc/api/objctests that still had the libjingle license header. BUG=webrtc:5418 TBR=tkchin@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1680293005 Cr-Commit-Position: refs/heads/master@{#11552}
92 lines
2.5 KiB
C++
92 lines
2.5 KiB
C++
/*
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* Copyright 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/api/audiotrack.h"
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#include "webrtc/base/checks.h"
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using rtc::scoped_refptr;
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namespace webrtc {
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const char MediaStreamTrackInterface::kAudioKind[] = "audio";
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// static
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scoped_refptr<AudioTrack> AudioTrack::Create(
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const std::string& id,
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const scoped_refptr<AudioSourceInterface>& source) {
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return new rtc::RefCountedObject<AudioTrack>(id, source);
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}
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AudioTrack::AudioTrack(const std::string& label,
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const scoped_refptr<AudioSourceInterface>& source)
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: MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
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if (audio_source_) {
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audio_source_->RegisterObserver(this);
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OnChanged();
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}
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}
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AudioTrack::~AudioTrack() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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set_state(MediaStreamTrackInterface::kEnded);
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if (audio_source_)
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audio_source_->UnregisterObserver(this);
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}
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std::string AudioTrack::kind() const {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return kAudioKind;
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}
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AudioSourceInterface* AudioTrack::GetSource() const {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return audio_source_.get();
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}
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void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (audio_source_)
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audio_source_->AddSink(sink);
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}
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void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (audio_source_)
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audio_source_->RemoveSink(sink);
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}
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void AudioTrack::OnChanged() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (state() == kFailed)
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return; // We can't recover from this state (do we ever set it?).
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TrackState new_state = kInitializing;
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// |audio_source_| must be non-null if we ever get here.
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switch (audio_source_->state()) {
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case MediaSourceInterface::kLive:
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case MediaSourceInterface::kMuted:
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new_state = kLive;
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break;
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case MediaSourceInterface::kEnded:
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new_state = kEnded;
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break;
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case MediaSourceInterface::kInitializing:
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default:
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// use kInitializing.
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break;
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}
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set_state(new_state);
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}
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} // namespace webrtc
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