| /* | 
 |  *  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 <utility> | 
 |  | 
 | #include "absl/algorithm/container.h" | 
 | #include "api/scoped_refptr.h" | 
 | #include "api/sequence_checker.h" | 
 | #include "api/task_queue/task_queue_base.h" | 
 | #include "rtc_base/checks.h" | 
 | #include "rtc_base/logging.h" | 
 | #include "rtc_base/strings/string_format.h" | 
 | #include "rtc_base/trace_event.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() = delete; | 
 |   AudioDataProxy(const AudioDataProxy&) = delete; | 
 |   AudioDataProxy& operator=(const AudioDataProxy&) = delete; | 
 |  | 
 |   ~AudioDataProxy() override { source_->OnAudioChannelGone(); } | 
 |  | 
 |   // AudioSinkInterface implementation. | 
 |   void OnData(const AudioSinkInterface::Data& audio) override { | 
 |     source_->OnData(audio); | 
 |   } | 
 |  | 
 |  private: | 
 |   const rtc::scoped_refptr<RemoteAudioSource> source_; | 
 | }; | 
 |  | 
 | RemoteAudioSource::RemoteAudioSource( | 
 |     TaskQueueBase* worker_thread, | 
 |     OnAudioChannelGoneAction on_audio_channel_gone_action) | 
 |     : main_thread_(TaskQueueBase::Current()), | 
 |       worker_thread_(worker_thread), | 
 |       on_audio_channel_gone_action_(on_audio_channel_gone_action), | 
 |       state_(MediaSourceInterface::kInitializing) { | 
 |   RTC_DCHECK(main_thread_); | 
 |   RTC_DCHECK(worker_thread_); | 
 | } | 
 |  | 
 | RemoteAudioSource::~RemoteAudioSource() { | 
 |   RTC_DCHECK(audio_observers_.empty()); | 
 |   if (!sinks_.empty()) { | 
 |     RTC_LOG(LS_WARNING) | 
 |         << "RemoteAudioSource destroyed while sinks_ is non-empty."; | 
 |   } | 
 | } | 
 |  | 
 | void RemoteAudioSource::Start( | 
 |     cricket::VoiceMediaReceiveChannelInterface* media_channel, | 
 |     absl::optional<uint32_t> ssrc) { | 
 |   RTC_DCHECK_RUN_ON(worker_thread_); | 
 |  | 
 |   // Register for callbacks immediately before AddSink so that we always get | 
 |   // notified when a channel goes out of scope (signaled when "AudioDataProxy" | 
 |   // is destroyed). | 
 |   RTC_DCHECK(media_channel); | 
 |   ssrc ? media_channel->SetRawAudioSink(*ssrc, | 
 |                                         std::make_unique<AudioDataProxy>(this)) | 
 |        : media_channel->SetDefaultRawAudioSink( | 
 |              std::make_unique<AudioDataProxy>(this)); | 
 | } | 
 |  | 
 | void RemoteAudioSource::Stop( | 
 |     cricket::VoiceMediaReceiveChannelInterface* media_channel, | 
 |     absl::optional<uint32_t> ssrc) { | 
 |   RTC_DCHECK_RUN_ON(worker_thread_); | 
 |   RTC_DCHECK(media_channel); | 
 |   ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr) | 
 |        : media_channel->SetDefaultRawAudioSink(nullptr); | 
 | } | 
 |  | 
 | void RemoteAudioSource::SetState(SourceState new_state) { | 
 |   RTC_DCHECK_RUN_ON(main_thread_); | 
 |   if (state_ != new_state) { | 
 |     state_ = new_state; | 
 |     FireOnChanged(); | 
 |   } | 
 | } | 
 |  | 
 | MediaSourceInterface::SourceState RemoteAudioSource::state() const { | 
 |   RTC_DCHECK_RUN_ON(main_thread_); | 
 |   return state_; | 
 | } | 
 |  | 
 | bool RemoteAudioSource::remote() const { | 
 |   RTC_DCHECK_RUN_ON(main_thread_); | 
 |   return true; | 
 | } | 
 |  | 
 | void RemoteAudioSource::SetVolume(double volume) { | 
 |   RTC_DCHECK_GE(volume, 0); | 
 |   RTC_DCHECK_LE(volume, 10); | 
 |   RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__, | 
 |                                         volume); | 
 |   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_RUN_ON(main_thread_); | 
 |   RTC_DCHECK(sink); | 
 |  | 
 |   MutexLock lock(&sink_lock_); | 
 |   RTC_DCHECK(!absl::c_linear_search(sinks_, sink)); | 
 |   sinks_.push_back(sink); | 
 | } | 
 |  | 
 | void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { | 
 |   RTC_DCHECK_RUN_ON(main_thread_); | 
 |   RTC_DCHECK(sink); | 
 |  | 
 |   MutexLock lock(&sink_lock_); | 
 |   sinks_.remove(sink); | 
 | } | 
 |  | 
 | void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { | 
 |   // Called on the externally-owned audio callback thread, via/from webrtc. | 
 |   TRACE_EVENT0("webrtc", "RemoteAudioSource::OnData"); | 
 |   MutexLock 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() { | 
 |   if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) { | 
 |     return; | 
 |   } | 
 |   // Called when the audio channel is deleted. It may be the worker thread or | 
 |   // may be a different task queue. | 
 |   // This object needs to live long enough for the cleanup logic in the posted | 
 |   // task to run, so take a reference to it. Sometimes the task may not be | 
 |   // processed (because the task queue was destroyed shortly after this call), | 
 |   // but that is fine because the task queue destructor will take care of | 
 |   // destroying task which will release the reference on RemoteAudioSource. | 
 |   rtc::scoped_refptr<RemoteAudioSource> thiz(this); | 
 |   main_thread_->PostTask([thiz = std::move(thiz)] { | 
 |     thiz->sinks_.clear(); | 
 |     thiz->SetState(MediaSourceInterface::kEnded); | 
 |   }); | 
 | } | 
 |  | 
 | }  // namespace webrtc |