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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, |
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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 explicit MessageHandler(RemoteAudioSource* source) : source_(source) {} |
| 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_IMPLICIT_CONSTRUCTORS(MessageHandler); |
| 55 }; |
| 56 |
| 57 class RemoteAudioSource::Sink : public AudioSinkInterface { |
| 58 public: |
| 59 explicit Sink(RemoteAudioSource* source) : source_(source) {} |
| 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_; |
| 69 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink); |
| 70 }; |
| 71 |
| 72 rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create( |
| 73 uint32_t ssrc, |
| 74 AudioProviderInterface* provider) { |
| 75 rtc::scoped_refptr<RemoteAudioSource> ret( |
| 76 new rtc::RefCountedObject<RemoteAudioSource>()); |
| 77 ret->Initialize(ssrc, provider); |
| 78 return ret; |
39 } | 79 } |
40 | 80 |
41 RemoteAudioSource::RemoteAudioSource() { | 81 RemoteAudioSource::RemoteAudioSource() |
| 82 : main_thread_(rtc::Thread::Current()), |
| 83 state_(MediaSourceInterface::kLive) { |
| 84 RTC_DCHECK(main_thread_); |
42 } | 85 } |
43 | 86 |
44 RemoteAudioSource::~RemoteAudioSource() { | 87 RemoteAudioSource::~RemoteAudioSource() { |
45 ASSERT(audio_observers_.empty()); | 88 RTC_DCHECK(main_thread_->IsCurrent()); |
| 89 RTC_DCHECK(audio_observers_.empty()); |
| 90 RTC_DCHECK(sinks_.empty()); |
| 91 } |
| 92 |
| 93 void RemoteAudioSource::Initialize(uint32_t ssrc, |
| 94 AudioProviderInterface* provider) { |
| 95 RTC_DCHECK(main_thread_->IsCurrent()); |
| 96 // To make sure we always get notified when the provider goes out of scope, |
| 97 // we register for callbacks here and not on demand in AddSink. |
| 98 if (provider) { // May be null in tests. |
| 99 provider->SetRawAudioSink( |
| 100 ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)))); |
| 101 } |
46 } | 102 } |
47 | 103 |
48 MediaSourceInterface::SourceState RemoteAudioSource::state() const { | 104 MediaSourceInterface::SourceState RemoteAudioSource::state() const { |
49 return MediaSourceInterface::kLive; | 105 RTC_DCHECK(main_thread_->IsCurrent()); |
| 106 return state_; |
50 } | 107 } |
51 | 108 |
52 void RemoteAudioSource::SetVolume(double volume) { | 109 void RemoteAudioSource::SetVolume(double volume) { |
53 ASSERT(volume >= 0 && volume <= 10); | 110 RTC_DCHECK(volume >= 0 && volume <= 10); |
54 for (AudioObserverList::iterator it = audio_observers_.begin(); | 111 for (auto* observer : audio_observers_) |
55 it != audio_observers_.end(); ++it) { | 112 observer->OnSetVolume(volume); |
56 (*it)->OnSetVolume(volume); | |
57 } | |
58 } | 113 } |
59 | 114 |
60 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { | 115 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { |
61 ASSERT(observer != NULL); | 116 RTC_DCHECK(observer != NULL); |
62 ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(), | 117 RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(), |
63 observer) == audio_observers_.end()); | 118 observer) == audio_observers_.end()); |
64 audio_observers_.push_back(observer); | 119 audio_observers_.push_back(observer); |
65 } | 120 } |
66 | 121 |
67 void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { | 122 void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { |
68 ASSERT(observer != NULL); | 123 RTC_DCHECK(observer != NULL); |
69 audio_observers_.remove(observer); | 124 audio_observers_.remove(observer); |
70 } | 125 } |
71 | 126 |
| 127 void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { |
| 128 RTC_DCHECK(main_thread_->IsCurrent()); |
| 129 RTC_DCHECK(sink); |
| 130 |
| 131 if (state_ != MediaSourceInterface::kLive) { |
| 132 LOG(LS_ERROR) << "Can't register sink as the source isn't live."; |
| 133 return; |
| 134 } |
| 135 |
| 136 rtc::CritScope lock(&sink_lock_); |
| 137 RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end()); |
| 138 sinks_.push_back(sink); |
| 139 } |
| 140 |
| 141 void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { |
| 142 RTC_DCHECK(main_thread_->IsCurrent()); |
| 143 RTC_DCHECK(sink); |
| 144 |
| 145 rtc::CritScope lock(&sink_lock_); |
| 146 sinks_.remove(sink); |
| 147 } |
| 148 |
| 149 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { |
| 150 // Called on the externally-owned audio callback thread, via/from webrtc. |
| 151 rtc::CritScope lock(&sink_lock_); |
| 152 for (auto* sink : sinks_) { |
| 153 sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, |
| 154 audio.samples_per_channel); |
| 155 } |
| 156 } |
| 157 |
| 158 void RemoteAudioSource::OnAudioProviderGone() { |
| 159 // Called when the data provider is deleted. It may be the worker thread |
| 160 // in libjingle or may be a different worker thread. |
| 161 main_thread_->Post(new MessageHandler(this)); |
| 162 } |
| 163 |
| 164 void RemoteAudioSource::OnMessage(rtc::Message* msg) { |
| 165 RTC_DCHECK(main_thread_->IsCurrent()); |
| 166 sinks_.clear(); |
| 167 state_ = MediaSourceInterface::kEnded; |
| 168 FireOnChanged(); |
| 169 } |
| 170 |
72 } // namespace webrtc | 171 } // namespace webrtc |
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