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| 1 /* | 1 /* | 
| 2 * libjingle | 2 * libjingle | 
| 3 * Copyright 2014 Google Inc. | 3 * Copyright 2014 Google Inc. | 
| 4 * | 4 * | 
| 5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without | 
| 6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: | 
| 7 * | 7 * | 
| 8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, | 
| 9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. | 
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 
| (...skipping 11 matching lines...) Expand all Loading... | |
| 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | 
| 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | 
| 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | 
| 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 
| 26 */ | 26 */ | 
| 27 | 27 | 
| 28 #include "talk/app/webrtc/remoteaudiosource.h" | 28 #include "talk/app/webrtc/remoteaudiosource.h" | 
| 29 | 29 | 
| 30 #include <algorithm> | 30 #include <algorithm> | 
| 31 #include <functional> | 31 #include <functional> | 
| 32 #include <utility> | |
| 32 | 33 | 
| 34 #include "talk/app/webrtc/mediastreamprovider.h" | |
| 35 #include "webrtc/base/checks.h" | |
| 33 #include "webrtc/base/logging.h" | 36 #include "webrtc/base/logging.h" | 
| 37 #include "webrtc/base/thread.h" | |
| 34 | 38 | 
| 35 namespace webrtc { | 39 namespace webrtc { | 
| 36 | 40 | 
| 37 rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() { | 41 class RemoteAudioSource::MessageHandler : public rtc::MessageHandler { | 
| 38 return new rtc::RefCountedObject<RemoteAudioSource>(); | 42 public: | 
| 43 MessageHandler(RemoteAudioSource* source) : source_(source) {} | |
| 
 
the sun
2015/12/11 19:46:28
explicit
 
tommi
2015/12/12 00:33:11
Done.
 
 | |
| 44 | |
| 45 private: | |
| 46 ~MessageHandler() override {} | |
| 47 | |
| 48 void OnMessage(rtc::Message* msg) override { | |
| 49 source_->OnMessage(msg); | |
| 50 delete this; | |
| 51 } | |
| 52 | |
| 53 const rtc::scoped_refptr<RemoteAudioSource> source_; | |
| 54 RTC_DISALLOW_COPY_AND_ASSIGN(MessageHandler); | |
| 
 
the sun
2015/12/11 19:46:29
DISALLOW_IMPLICIT...
 
tommi
2015/12/12 00:33:11
Done.
 
 | |
| 55 }; | |
| 56 | |
| 57 class RemoteAudioSource::Sink : public AudioSinkInterface { | |
| 58 public: | |
| 59 Sink(RemoteAudioSource* source) : source_(source) {} | |
| 
 
the sun
2015/12/11 19:46:28
explicit
 
tommi
2015/12/12 00:33:11
Done.
 
 | |
| 60 ~Sink() override { source_->OnAudioProviderGone(); } | |
| 61 | |
| 62 private: | |
| 63 void OnData(const AudioSinkInterface::Data& audio) override { | |
| 64 if (source_) | |
| 65 source_->OnData(audio); | |
| 66 } | |
| 67 | |
| 68 const rtc::scoped_refptr<RemoteAudioSource> source_; | |
| 
 
the sun
2015/12/11 19:46:28
DISALLOW_IMPLICIT...
 
tommi
2015/12/12 00:33:11
Done.
 
 | |
| 69 }; | |
| 70 | |
| 71 rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create( | |
| 72 uint32_t ssrc, | |
| 73 AudioProviderInterface* provider) { | |
| 74 rtc::scoped_refptr<RemoteAudioSource> ret( | |
| 75 new rtc::RefCountedObject<RemoteAudioSource>()); | |
| 76 ret->Initialize(ssrc, provider); | |
| 77 return ret; | |
| 39 } | 78 } | 
| 40 | 79 | 
| 41 RemoteAudioSource::RemoteAudioSource() { | 80 RemoteAudioSource::RemoteAudioSource() | 
| 81 : main_thread_(rtc::Thread::Current()), | |
| 82 state_(MediaSourceInterface::kLive) { | |
| 83 RTC_DCHECK(main_thread_); | |
| 42 } | 84 } | 
| 43 | 85 | 
| 44 RemoteAudioSource::~RemoteAudioSource() { | 86 RemoteAudioSource::~RemoteAudioSource() { | 
| 45 ASSERT(audio_observers_.empty()); | 87 RTC_DCHECK(main_thread_->IsCurrent()); | 
| 88 RTC_DCHECK(audio_observers_.empty()); | |
| 89 RTC_DCHECK(sinks_.empty()); | |
| 90 } | |
| 91 | |
| 92 void RemoteAudioSource::Initialize(uint32_t ssrc, | |
| 93 AudioProviderInterface* provider) { | |
| 94 RTC_DCHECK(main_thread_->IsCurrent()); | |
| 95 // To make sure we always get notified when the provider goes out of scope, | |
| 96 // we register for callbacks here and not on demand in AddSink. | |
| 97 if (provider) { // May be null in tests. | |
| 98 provider->SetRawAudioSink( | |
| 99 ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)))); | |
| 100 } | |
| 46 } | 101 } | 
| 47 | 102 | 
| 48 MediaSourceInterface::SourceState RemoteAudioSource::state() const { | 103 MediaSourceInterface::SourceState RemoteAudioSource::state() const { | 
| 49 return MediaSourceInterface::kLive; | 104 RTC_DCHECK(main_thread_->IsCurrent()); | 
| 105 return state_; | |
| 50 } | 106 } | 
| 51 | 107 | 
| 52 void RemoteAudioSource::SetVolume(double volume) { | 108 void RemoteAudioSource::SetVolume(double volume) { | 
| 53 ASSERT(volume >= 0 && volume <= 10); | 109 RTC_DCHECK(volume >= 0 && volume <= 10); | 
| 54 for (AudioObserverList::iterator it = audio_observers_.begin(); | 110 for (auto* observer : audio_observers_) | 
| 55 it != audio_observers_.end(); ++it) { | 111 observer->OnSetVolume(volume); | 
| 56 (*it)->OnSetVolume(volume); | |
| 57 } | |
| 58 } | 112 } | 
| 59 | 113 | 
| 60 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { | 114 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { | 
| 61 ASSERT(observer != NULL); | 115 RTC_DCHECK(observer != NULL); | 
| 62 ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(), | 116 RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(), | 
| 63 observer) == audio_observers_.end()); | 117 observer) == audio_observers_.end()); | 
| 64 audio_observers_.push_back(observer); | 118 audio_observers_.push_back(observer); | 
| 65 } | 119 } | 
| 66 | 120 | 
| 67 void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { | 121 void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { | 
| 68 ASSERT(observer != NULL); | 122 RTC_DCHECK(observer != NULL); | 
| 69 audio_observers_.remove(observer); | 123 audio_observers_.remove(observer); | 
| 70 } | 124 } | 
| 71 | 125 | 
| 126 void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { | |
| 127 RTC_DCHECK(main_thread_->IsCurrent()); | |
| 128 RTC_DCHECK(sink); | |
| 129 | |
| 130 if (state_ != MediaSourceInterface::kLive) { | |
| 131 LOG(LS_ERROR) << "Can't register sink as the source isn't live."; | |
| 132 return; | |
| 133 } | |
| 134 | |
| 135 rtc::CritScope lock(&sink_lock_); | |
| 136 RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end()); | |
| 137 sinks_.push_back(sink); | |
| 138 } | |
| 139 | |
| 140 void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { | |
| 141 RTC_DCHECK(main_thread_->IsCurrent()); | |
| 142 RTC_DCHECK(sink); | |
| 143 | |
| 144 rtc::CritScope lock(&sink_lock_); | |
| 145 sinks_.remove(sink); | |
| 146 } | |
| 147 | |
| 148 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { | |
| 149 // Called on the externally-owned audio callback thread, via/from webrtc. | |
| 150 rtc::CritScope lock(&sink_lock_); | |
| 151 for (auto* sink : sinks_) { | |
| 152 sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, | |
| 153 audio.samples_per_channel); | |
| 154 } | |
| 155 } | |
| 156 | |
| 157 void RemoteAudioSource::OnAudioProviderGone() { | |
| 158 // Called when the data provider is deleted. It may be the worker thread | |
| 159 // in libjingle or may be a different worker thread. | |
| 160 main_thread_->Post(new MessageHandler(this)); | |
| 161 } | |
| 162 | |
| 163 void RemoteAudioSource::OnMessage(rtc::Message* msg) { | |
| 164 RTC_DCHECK(main_thread_->IsCurrent()); | |
| 165 sinks_.clear(); | |
| 166 state_ = MediaSourceInterface::kEnded; | |
| 167 FireOnChanged(); | |
| 168 } | |
| 169 | |
| 72 } // namespace webrtc | 170 } // namespace webrtc | 
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