|  | /* | 
|  | *  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/remote_audio_source.h" | 
|  |  | 
|  | #include <stddef.h> | 
|  |  | 
|  | #include <memory> | 
|  | #include <string> | 
|  |  | 
|  | #include "absl/algorithm/container.h" | 
|  | #include "api/scoped_refptr.h" | 
|  | #include "rtc_base/checks.h" | 
|  | #include "rtc_base/constructor_magic.h" | 
|  | #include "rtc_base/location.h" | 
|  | #include "rtc_base/logging.h" | 
|  | #include "rtc_base/numerics/safe_conversions.h" | 
|  | #include "rtc_base/thread.h" | 
|  | #include "rtc_base/thread_checker.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | // This proxy is passed to the underlying media engine to receive audio data as | 
|  | // they come in. The data will then be passed back up to the RemoteAudioSource | 
|  | // which will fan it out to all the sinks that have been added to it. | 
|  | class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface { | 
|  | public: | 
|  | explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) { | 
|  | RTC_DCHECK(source); | 
|  | } | 
|  | ~AudioDataProxy() override { source_->OnAudioChannelGone(); } | 
|  |  | 
|  | // AudioSinkInterface implementation. | 
|  | void OnData(const AudioSinkInterface::Data& audio) override { | 
|  | source_->OnData(audio); | 
|  | } | 
|  |  | 
|  | private: | 
|  | const rtc::scoped_refptr<RemoteAudioSource> source_; | 
|  |  | 
|  | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy); | 
|  | }; | 
|  |  | 
|  | RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread) | 
|  | : main_thread_(rtc::Thread::Current()), | 
|  | worker_thread_(worker_thread), | 
|  | state_(MediaSourceInterface::kLive) { | 
|  | RTC_DCHECK(main_thread_); | 
|  | RTC_DCHECK(worker_thread_); | 
|  | } | 
|  |  | 
|  | RemoteAudioSource::~RemoteAudioSource() { | 
|  | RTC_DCHECK(main_thread_->IsCurrent()); | 
|  | RTC_DCHECK(audio_observers_.empty()); | 
|  | RTC_DCHECK(sinks_.empty()); | 
|  | } | 
|  |  | 
|  | void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel, | 
|  | absl::optional<uint32_t> ssrc) { | 
|  | RTC_DCHECK_RUN_ON(main_thread_); | 
|  | RTC_DCHECK(media_channel); | 
|  |  | 
|  | // Register for callbacks immediately before AddSink so that we always get | 
|  | // notified when a channel goes out of scope (signaled when "AudioDataProxy" | 
|  | // is destroyed). | 
|  | worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { | 
|  | ssrc ? media_channel->SetRawAudioSink( | 
|  | *ssrc, std::make_unique<AudioDataProxy>(this)) | 
|  | : media_channel->SetDefaultRawAudioSink( | 
|  | std::make_unique<AudioDataProxy>(this)); | 
|  | }); | 
|  | } | 
|  |  | 
|  | void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel, | 
|  | absl::optional<uint32_t> ssrc) { | 
|  | RTC_DCHECK_RUN_ON(main_thread_); | 
|  | RTC_DCHECK(media_channel); | 
|  |  | 
|  | worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { | 
|  | ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr) | 
|  | : media_channel->SetDefaultRawAudioSink(nullptr); | 
|  | }); | 
|  | } | 
|  |  | 
|  | 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(!absl::c_linear_search(audio_observers_, observer)); | 
|  | 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(!absl::c_linear_search(sinks_, sink)); | 
|  | 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_) { | 
|  | // When peerconnection acts as an audio source, it should not provide | 
|  | // absolute capture timestamp. | 
|  | sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, | 
|  | audio.samples_per_channel, | 
|  | /*absolute_capture_timestamp_ms=*/absl::nullopt); | 
|  | } | 
|  | } | 
|  |  | 
|  | 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 |