
This CL has been generated with the following script: for m in PLOG \ LOG_TAG \ LOG_GLEM \ LOG_GLE_EX \ LOG_GLE \ LAST_SYSTEM_ERROR \ LOG_ERRNO_EX \ LOG_ERRNO \ LOG_ERR_EX \ LOG_ERR \ LOG_V \ LOG_F \ LOG_T_F \ LOG_E \ LOG_T \ LOG_CHECK_LEVEL_V \ LOG_CHECK_LEVEL \ LOG do git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g" done git checkout rtc_base/logging.h git cl format Bug: webrtc:8452 Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600 Reviewed-on: https://webrtc-review.googlesource.com/21325 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20617}
155 lines
4.8 KiB
C++
155 lines
4.8 KiB
C++
/*
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* Copyright 2014 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/remoteaudiosource.h"
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#include <algorithm>
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#include <functional>
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#include <memory>
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#include <utility>
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#include "rtc_base/checks.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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class RemoteAudioSource::Sink : public AudioSinkInterface {
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public:
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explicit Sink(RemoteAudioSource* source) : source_(source) {}
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~Sink() override { source_->OnAudioChannelGone(); }
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private:
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void OnData(const AudioSinkInterface::Data& audio) override {
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if (source_)
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source_->OnData(audio);
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}
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const rtc::scoped_refptr<RemoteAudioSource> source_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
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};
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rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
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uint32_t ssrc,
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cricket::VoiceChannel* channel) {
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rtc::scoped_refptr<RemoteAudioSource> ret(
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new rtc::RefCountedObject<RemoteAudioSource>());
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ret->Initialize(ssrc, channel);
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return ret;
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}
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RemoteAudioSource::RemoteAudioSource()
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: main_thread_(rtc::Thread::Current()),
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state_(MediaSourceInterface::kLive) {
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RTC_DCHECK(main_thread_);
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}
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RemoteAudioSource::~RemoteAudioSource() {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(audio_observers_.empty());
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RTC_DCHECK(sinks_.empty());
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}
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void RemoteAudioSource::Initialize(uint32_t ssrc,
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cricket::VoiceChannel* channel) {
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RTC_DCHECK(main_thread_->IsCurrent());
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// To make sure we always get notified when the channel goes out of scope,
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// we register for callbacks here and not on demand in AddSink.
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if (channel) { // May be null in tests.
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channel->SetRawAudioSink(
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ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
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}
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}
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MediaSourceInterface::SourceState RemoteAudioSource::state() const {
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RTC_DCHECK(main_thread_->IsCurrent());
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return state_;
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}
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bool RemoteAudioSource::remote() const {
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RTC_DCHECK(main_thread_->IsCurrent());
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return true;
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}
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void RemoteAudioSource::SetVolume(double volume) {
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RTC_DCHECK_GE(volume, 0);
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RTC_DCHECK_LE(volume, 10);
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for (auto* observer : audio_observers_)
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observer->OnSetVolume(volume);
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}
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void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
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RTC_DCHECK(observer != NULL);
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RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(),
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observer) == audio_observers_.end());
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audio_observers_.push_back(observer);
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}
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void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
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RTC_DCHECK(observer != NULL);
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audio_observers_.remove(observer);
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}
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void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(sink);
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if (state_ != MediaSourceInterface::kLive) {
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RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
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return;
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}
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rtc::CritScope lock(&sink_lock_);
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RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
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sinks_.push_back(sink);
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}
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void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(main_thread_->IsCurrent());
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RTC_DCHECK(sink);
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rtc::CritScope lock(&sink_lock_);
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sinks_.remove(sink);
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}
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void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
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// Called on the externally-owned audio callback thread, via/from webrtc.
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rtc::CritScope lock(&sink_lock_);
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for (auto* sink : sinks_) {
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sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
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audio.samples_per_channel);
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}
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}
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void RemoteAudioSource::OnAudioChannelGone() {
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// Called when the audio channel is deleted. It may be the worker thread
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// in libjingle or may be a different worker thread.
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// This object needs to live long enough for the cleanup logic in OnMessage to
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// run, so take a reference to it as the data. Sometimes the message may not
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// be processed (because the thread was destroyed shortly after this call),
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// but that is fine because the thread destructor will take care of destroying
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// the message data which will release the reference on RemoteAudioSource.
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main_thread_->Post(RTC_FROM_HERE, this, 0,
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new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
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}
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void RemoteAudioSource::OnMessage(rtc::Message* msg) {
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RTC_DCHECK(main_thread_->IsCurrent());
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sinks_.clear();
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state_ = MediaSourceInterface::kEnded;
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FireOnChanged();
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// Will possibly delete this RemoteAudioSource since it is reference counted
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// in the message.
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delete msg->pdata;
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}
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} // namespace webrtc
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