| /* |
| * 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 "api/sequence_checker.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/string_format.h" |
| #include "rtc_base/thread.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( |
| rtc::Thread* worker_thread, |
| OnAudioChannelGoneAction on_audio_channel_gone_action) |
| : main_thread_(rtc::Thread::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::VoiceMediaChannel* 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::VoiceMediaChannel* 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. |
| 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 |
| // 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_RUN_ON(main_thread_); |
| sinks_.clear(); |
| SetState(MediaSourceInterface::kEnded); |
| // Will possibly delete this RemoteAudioSource since it is reference counted |
| // in the message. |
| delete msg->pdata; |
| } |
| |
| } // namespace webrtc |