1 /*
2  *  Copyright 2014 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #include "pc/remote_audio_source.h"
12 
13 #include <stddef.h>
14 
15 #include <memory>
16 
17 #include "absl/algorithm/container.h"
18 #include "api/scoped_refptr.h"
19 #include "api/sequence_checker.h"
20 #include "rtc_base/checks.h"
21 #include "rtc_base/location.h"
22 #include "rtc_base/logging.h"
23 #include "rtc_base/strings/string_format.h"
24 #include "rtc_base/thread.h"
25 
26 namespace webrtc {
27 
28 // This proxy is passed to the underlying media engine to receive audio data as
29 // they come in. The data will then be passed back up to the RemoteAudioSource
30 // which will fan it out to all the sinks that have been added to it.
31 class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
32  public:
AudioDataProxy(RemoteAudioSource * source)33   explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
34     RTC_DCHECK(source);
35   }
36 
37   AudioDataProxy() = delete;
38   AudioDataProxy(const AudioDataProxy&) = delete;
39   AudioDataProxy& operator=(const AudioDataProxy&) = delete;
40 
~AudioDataProxy()41   ~AudioDataProxy() override { source_->OnAudioChannelGone(); }
42 
43   // AudioSinkInterface implementation.
OnData(const AudioSinkInterface::Data & audio)44   void OnData(const AudioSinkInterface::Data& audio) override {
45     source_->OnData(audio);
46   }
47 
48  private:
49   const rtc::scoped_refptr<RemoteAudioSource> source_;
50 };
51 
RemoteAudioSource(rtc::Thread * worker_thread,OnAudioChannelGoneAction on_audio_channel_gone_action)52 RemoteAudioSource::RemoteAudioSource(
53     rtc::Thread* worker_thread,
54     OnAudioChannelGoneAction on_audio_channel_gone_action)
55     : main_thread_(rtc::Thread::Current()),
56       worker_thread_(worker_thread),
57       on_audio_channel_gone_action_(on_audio_channel_gone_action),
58       state_(MediaSourceInterface::kLive) {
59   RTC_DCHECK(main_thread_);
60   RTC_DCHECK(worker_thread_);
61 }
62 
~RemoteAudioSource()63 RemoteAudioSource::~RemoteAudioSource() {
64   RTC_DCHECK(main_thread_->IsCurrent());
65   RTC_DCHECK(audio_observers_.empty());
66   RTC_DCHECK(sinks_.empty());
67 }
68 
Start(cricket::VoiceMediaChannel * media_channel,absl::optional<uint32_t> ssrc)69 void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
70                               absl::optional<uint32_t> ssrc) {
71   RTC_DCHECK_RUN_ON(main_thread_);
72   RTC_DCHECK(media_channel);
73 
74   // Register for callbacks immediately before AddSink so that we always get
75   // notified when a channel goes out of scope (signaled when "AudioDataProxy"
76   // is destroyed).
77   worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
78     ssrc ? media_channel->SetRawAudioSink(
79                *ssrc, std::make_unique<AudioDataProxy>(this))
80          : media_channel->SetDefaultRawAudioSink(
81                std::make_unique<AudioDataProxy>(this));
82   });
83 }
84 
Stop(cricket::VoiceMediaChannel * media_channel,absl::optional<uint32_t> ssrc)85 void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
86                              absl::optional<uint32_t> ssrc) {
87   RTC_DCHECK_RUN_ON(main_thread_);
88   RTC_DCHECK(media_channel);
89 
90   worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
91     ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
92          : media_channel->SetDefaultRawAudioSink(nullptr);
93   });
94 }
95 
SetState(SourceState new_state)96 void RemoteAudioSource::SetState(SourceState new_state) {
97   if (state_ != new_state) {
98     state_ = new_state;
99     FireOnChanged();
100   }
101 }
102 
state() const103 MediaSourceInterface::SourceState RemoteAudioSource::state() const {
104   RTC_DCHECK(main_thread_->IsCurrent());
105   return state_;
106 }
107 
remote() const108 bool RemoteAudioSource::remote() const {
109   RTC_DCHECK(main_thread_->IsCurrent());
110   return true;
111 }
112 
SetVolume(double volume)113 void RemoteAudioSource::SetVolume(double volume) {
114   RTC_DCHECK_GE(volume, 0);
115   RTC_DCHECK_LE(volume, 10);
116   RTC_LOG(LS_INFO) << rtc::StringFormat("RAS::%s({volume=%.2f})", __func__,
117                                         volume);
118   for (auto* observer : audio_observers_) {
119     observer->OnSetVolume(volume);
120   }
121 }
122 
RegisterAudioObserver(AudioObserver * observer)123 void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
124   RTC_DCHECK(observer != NULL);
125   RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
126   audio_observers_.push_back(observer);
127 }
128 
UnregisterAudioObserver(AudioObserver * observer)129 void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
130   RTC_DCHECK(observer != NULL);
131   audio_observers_.remove(observer);
132 }
133 
AddSink(AudioTrackSinkInterface * sink)134 void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
135   RTC_DCHECK(main_thread_->IsCurrent());
136   RTC_DCHECK(sink);
137 
138   if (state_ != MediaSourceInterface::kLive) {
139     RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
140     return;
141   }
142 
143   MutexLock lock(&sink_lock_);
144   RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
145   sinks_.push_back(sink);
146 }
147 
RemoveSink(AudioTrackSinkInterface * sink)148 void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
149   RTC_DCHECK(main_thread_->IsCurrent());
150   RTC_DCHECK(sink);
151 
152   MutexLock lock(&sink_lock_);
153   sinks_.remove(sink);
154 }
155 
OnData(const AudioSinkInterface::Data & audio)156 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
157   // Called on the externally-owned audio callback thread, via/from webrtc.
158   MutexLock lock(&sink_lock_);
159   for (auto* sink : sinks_) {
160     // When peerconnection acts as an audio source, it should not provide
161     // absolute capture timestamp.
162     sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
163                  audio.samples_per_channel,
164                  /*absolute_capture_timestamp_ms=*/absl::nullopt);
165   }
166 }
167 
OnAudioChannelGone()168 void RemoteAudioSource::OnAudioChannelGone() {
169   if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
170     return;
171   }
172   // Called when the audio channel is deleted.  It may be the worker thread
173   // in libjingle or may be a different worker thread.
174   // This object needs to live long enough for the cleanup logic in OnMessage to
175   // run, so take a reference to it as the data. Sometimes the message may not
176   // be processed (because the thread was destroyed shortly after this call),
177   // but that is fine because the thread destructor will take care of destroying
178   // the message data which will release the reference on RemoteAudioSource.
179   main_thread_->Post(RTC_FROM_HERE, this, 0,
180                      new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
181 }
182 
OnMessage(rtc::Message * msg)183 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
184   RTC_DCHECK(main_thread_->IsCurrent());
185   sinks_.clear();
186   SetState(MediaSourceInterface::kEnded);
187   // Will possibly delete this RemoteAudioSource since it is reference counted
188   // in the message.
189   delete msg->pdata;
190 }
191 
192 }  // namespace webrtc
193