| /* |
| * Copyright 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/remoteaudiosource.h" |
| |
| #include <algorithm> |
| #include <functional> |
| #include <memory> |
| #include <utility> |
| |
| #include "rtc_base/checks.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/thread.h" |
| |
| namespace webrtc { |
| |
| class RemoteAudioSource::Sink : public AudioSinkInterface { |
| public: |
| explicit Sink(RemoteAudioSource* source) : source_(source) {} |
| ~Sink() override { source_->OnAudioChannelGone(); } |
| |
| private: |
| void OnData(const AudioSinkInterface::Data& audio) override { |
| if (source_) |
| source_->OnData(audio); |
| } |
| |
| const rtc::scoped_refptr<RemoteAudioSource> source_; |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink); |
| }; |
| |
| rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create( |
| uint32_t ssrc, |
| cricket::VoiceChannel* channel) { |
| rtc::scoped_refptr<RemoteAudioSource> ret( |
| new rtc::RefCountedObject<RemoteAudioSource>()); |
| ret->Initialize(ssrc, channel); |
| return ret; |
| } |
| |
| RemoteAudioSource::RemoteAudioSource() |
| : main_thread_(rtc::Thread::Current()), |
| state_(MediaSourceInterface::kLive) { |
| RTC_DCHECK(main_thread_); |
| } |
| |
| RemoteAudioSource::~RemoteAudioSource() { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| RTC_DCHECK(audio_observers_.empty()); |
| RTC_DCHECK(sinks_.empty()); |
| } |
| |
| void RemoteAudioSource::Initialize(uint32_t ssrc, |
| cricket::VoiceChannel* channel) { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| // To make sure we always get notified when the channel goes out of scope, |
| // we register for callbacks here and not on demand in AddSink. |
| if (channel) { // May be null in tests. |
| channel->SetRawAudioSink( |
| ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this))); |
| } |
| } |
| |
| MediaSourceInterface::SourceState RemoteAudioSource::state() const { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| return state_; |
| } |
| |
| bool RemoteAudioSource::remote() const { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| return true; |
| } |
| |
| void RemoteAudioSource::SetVolume(double volume) { |
| RTC_DCHECK_GE(volume, 0); |
| RTC_DCHECK_LE(volume, 10); |
| for (auto* observer : audio_observers_) |
| observer->OnSetVolume(volume); |
| } |
| |
| void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { |
| RTC_DCHECK(observer != NULL); |
| RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(), |
| observer) == audio_observers_.end()); |
| audio_observers_.push_back(observer); |
| } |
| |
| void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { |
| RTC_DCHECK(observer != NULL); |
| audio_observers_.remove(observer); |
| } |
| |
| void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| RTC_DCHECK(sink); |
| |
| if (state_ != MediaSourceInterface::kLive) { |
| RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live."; |
| return; |
| } |
| |
| rtc::CritScope lock(&sink_lock_); |
| RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end()); |
| sinks_.push_back(sink); |
| } |
| |
| void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| RTC_DCHECK(sink); |
| |
| rtc::CritScope lock(&sink_lock_); |
| sinks_.remove(sink); |
| } |
| |
| void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { |
| // Called on the externally-owned audio callback thread, via/from webrtc. |
| rtc::CritScope lock(&sink_lock_); |
| for (auto* sink : sinks_) { |
| sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, |
| audio.samples_per_channel); |
| } |
| } |
| |
| void RemoteAudioSource::OnAudioChannelGone() { |
| // Called when the audio channel is deleted. It may be the worker thread |
| // in libjingle or may be a different worker thread. |
| // This object needs to live long enough for the cleanup logic in OnMessage to |
| // run, so take a reference to it as the data. Sometimes the message may not |
| // be processed (because the thread was destroyed shortly after this call), |
| // but that is fine because the thread destructor will take care of destroying |
| // the message data which will release the reference on RemoteAudioSource. |
| main_thread_->Post(RTC_FROM_HERE, this, 0, |
| new rtc::ScopedRefMessageData<RemoteAudioSource>(this)); |
| } |
| |
| void RemoteAudioSource::OnMessage(rtc::Message* msg) { |
| RTC_DCHECK(main_thread_->IsCurrent()); |
| sinks_.clear(); |
| state_ = MediaSourceInterface::kEnded; |
| FireOnChanged(); |
| // Will possibly delete this RemoteAudioSource since it is reference counted |
| // in the message. |
| delete msg->pdata; |
| } |
| |
| } // namespace webrtc |