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Side by Side Diff: talk/app/webrtc/remoteaudiosource.cc

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years ago
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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
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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