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author | Tommi <tommi@webrtc.org> | 2015-12-12 01:37:01 +0100 |
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committer | Tommi <tommi@webrtc.org> | 2015-12-12 00:37:14 +0000 |
commit | f888bb58da04c5095759b5ec7ce2e1fa2cd414fd (patch) | |
tree | 2cd9d8f7b8d734c018d5886b18ba82ade2fa339a /talk/app/webrtc/remoteaudiosource.cc | |
parent | f67c548576ad957a1e9c3196e11d45f41e320424 (diff) | |
download | webrtc-f888bb58da04c5095759b5ec7ce2e1fa2cd414fd.tar.gz |
Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1505253004 .
Cr-Commit-Position: refs/heads/master@{#10995}
Diffstat (limited to 'talk/app/webrtc/remoteaudiosource.cc')
-rw-r--r-- | talk/app/webrtc/remoteaudiosource.cc | 127 |
1 files changed, 113 insertions, 14 deletions
diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc index 41f3d8798a..e5af1e9487 100644 --- a/talk/app/webrtc/remoteaudiosource.cc +++ b/talk/app/webrtc/remoteaudiosource.cc @@ -29,44 +29,143 @@ #include <algorithm> #include <functional> +#include <utility> +#include "talk/app/webrtc/mediastreamprovider.h" +#include "webrtc/base/checks.h" #include "webrtc/base/logging.h" +#include "webrtc/base/thread.h" namespace webrtc { -rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() { - return new rtc::RefCountedObject<RemoteAudioSource>(); +class RemoteAudioSource::MessageHandler : public rtc::MessageHandler { + public: + explicit MessageHandler(RemoteAudioSource* source) : source_(source) {} + + private: + ~MessageHandler() override {} + + void OnMessage(rtc::Message* msg) override { + source_->OnMessage(msg); + delete this; + } + + const rtc::scoped_refptr<RemoteAudioSource> source_; + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler); +}; + +class RemoteAudioSource::Sink : public AudioSinkInterface { + public: + explicit Sink(RemoteAudioSource* source) : source_(source) {} + ~Sink() override { source_->OnAudioProviderGone(); } + + 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, + AudioProviderInterface* provider) { + rtc::scoped_refptr<RemoteAudioSource> ret( + new rtc::RefCountedObject<RemoteAudioSource>()); + ret->Initialize(ssrc, provider); + return ret; } -RemoteAudioSource::RemoteAudioSource() { +RemoteAudioSource::RemoteAudioSource() + : main_thread_(rtc::Thread::Current()), + state_(MediaSourceInterface::kLive) { + RTC_DCHECK(main_thread_); } RemoteAudioSource::~RemoteAudioSource() { - ASSERT(audio_observers_.empty()); + RTC_DCHECK(main_thread_->IsCurrent()); + RTC_DCHECK(audio_observers_.empty()); + RTC_DCHECK(sinks_.empty()); +} + +void RemoteAudioSource::Initialize(uint32_t ssrc, + AudioProviderInterface* provider) { + RTC_DCHECK(main_thread_->IsCurrent()); + // To make sure we always get notified when the provider goes out of scope, + // we register for callbacks here and not on demand in AddSink. + if (provider) { // May be null in tests. + provider->SetRawAudioSink( + ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)))); + } } MediaSourceInterface::SourceState RemoteAudioSource::state() const { - return MediaSourceInterface::kLive; + RTC_DCHECK(main_thread_->IsCurrent()); + return state_; } void RemoteAudioSource::SetVolume(double volume) { - ASSERT(volume >= 0 && volume <= 10); - for (AudioObserverList::iterator it = audio_observers_.begin(); - it != audio_observers_.end(); ++it) { - (*it)->OnSetVolume(volume); - } + RTC_DCHECK(volume >= 0 && volume <= 10); + for (auto* observer : audio_observers_) + observer->OnSetVolume(volume); } void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { - ASSERT(observer != NULL); - ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(), - observer) == audio_observers_.end()); + 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) { - ASSERT(observer != NULL); + 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) { + 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::OnAudioProviderGone() { + // Called when the data provider is deleted. It may be the worker thread + // in libjingle or may be a different worker thread. + main_thread_->Post(new MessageHandler(this)); +} + +void RemoteAudioSource::OnMessage(rtc::Message* msg) { + RTC_DCHECK(main_thread_->IsCurrent()); + sinks_.clear(); + state_ = MediaSourceInterface::kEnded; + FireOnChanged(); +} + } // namespace webrtc |