d25fa78daf
This reverts commit b7239a9dc825ddb25dbc800aed3a065163b9a10e. Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h. Original change's description: > Make rtc_base/refcount.h self contained, not including refcountedobject.h. > > The refcount.h file doesn't depend on anything from > refcountedobject.h. The motivation of this change to make it possible > to add additional declarations to refcount.h, and include it from > refcountedobject.h. > > Bug: webrtc:8270 > Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373 > Reviewed-on: https://webrtc-review.googlesource.com/5760 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20106} TBR=kwiberg@webrtc.org,nisse@webrtc.org Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8270 Reviewed-on: https://webrtc-review.googlesource.com/5800 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20107}
74 lines
2.0 KiB
C++
74 lines
2.0 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 "pc/audiotrack.h"
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#include "rtc_base/checks.h"
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using rtc::scoped_refptr;
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namespace webrtc {
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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 (audio_source_->state() == MediaSourceInterface::kEnded) {
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set_state(kEnded);
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} else {
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set_state(kLive);
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}
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}
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} // namespace webrtc
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