wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 1 | /* |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 2 | * Copyright 2014 The WebRTC project authors. All Rights Reserved. |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 3 | * |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 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. |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 9 | */ |
| 10 | |
Henrik Kjellander | 15583c1 | 2016-02-10 10:53:12 +0100 | [diff] [blame] | 11 | #include "webrtc/api/remoteaudiosource.h" |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 12 | |
| 13 | #include <algorithm> |
| 14 | #include <functional> |
kwiberg | d1fe281 | 2016-04-27 06:47:29 -0700 | [diff] [blame] | 15 | #include <memory> |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 16 | #include <utility> |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 17 | |
Henrik Kjellander | 15583c1 | 2016-02-10 10:53:12 +0100 | [diff] [blame] | 18 | #include "webrtc/api/mediastreamprovider.h" |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 19 | #include "webrtc/base/checks.h" |
kwiberg | 4485ffb | 2016-04-26 08:14:39 -0700 | [diff] [blame] | 20 | #include "webrtc/base/constructormagic.h" |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 21 | #include "webrtc/base/logging.h" |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 22 | #include "webrtc/base/thread.h" |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 23 | |
| 24 | namespace webrtc { |
| 25 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 26 | class RemoteAudioSource::MessageHandler : public rtc::MessageHandler { |
| 27 | public: |
| 28 | explicit MessageHandler(RemoteAudioSource* source) : source_(source) {} |
| 29 | |
| 30 | private: |
| 31 | ~MessageHandler() override {} |
| 32 | |
| 33 | void OnMessage(rtc::Message* msg) override { |
| 34 | source_->OnMessage(msg); |
| 35 | delete this; |
| 36 | } |
| 37 | |
| 38 | const rtc::scoped_refptr<RemoteAudioSource> source_; |
| 39 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler); |
| 40 | }; |
| 41 | |
| 42 | class RemoteAudioSource::Sink : public AudioSinkInterface { |
| 43 | public: |
| 44 | explicit Sink(RemoteAudioSource* source) : source_(source) {} |
| 45 | ~Sink() override { source_->OnAudioProviderGone(); } |
| 46 | |
| 47 | private: |
| 48 | void OnData(const AudioSinkInterface::Data& audio) override { |
| 49 | if (source_) |
| 50 | source_->OnData(audio); |
| 51 | } |
| 52 | |
| 53 | const rtc::scoped_refptr<RemoteAudioSource> source_; |
| 54 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink); |
| 55 | }; |
| 56 | |
| 57 | rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create( |
| 58 | uint32_t ssrc, |
| 59 | AudioProviderInterface* provider) { |
| 60 | rtc::scoped_refptr<RemoteAudioSource> ret( |
| 61 | new rtc::RefCountedObject<RemoteAudioSource>()); |
| 62 | ret->Initialize(ssrc, provider); |
| 63 | return ret; |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 64 | } |
| 65 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 66 | RemoteAudioSource::RemoteAudioSource() |
| 67 | : main_thread_(rtc::Thread::Current()), |
| 68 | state_(MediaSourceInterface::kLive) { |
| 69 | RTC_DCHECK(main_thread_); |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 70 | } |
| 71 | |
| 72 | RemoteAudioSource::~RemoteAudioSource() { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 73 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 74 | RTC_DCHECK(audio_observers_.empty()); |
| 75 | RTC_DCHECK(sinks_.empty()); |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 76 | } |
| 77 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 78 | void RemoteAudioSource::Initialize(uint32_t ssrc, |
| 79 | AudioProviderInterface* provider) { |
| 80 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 81 | // To make sure we always get notified when the provider goes out of scope, |
| 82 | // we register for callbacks here and not on demand in AddSink. |
| 83 | if (provider) { // May be null in tests. |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 84 | provider->SetRawAudioSink( |
kwiberg | d1fe281 | 2016-04-27 06:47:29 -0700 | [diff] [blame] | 85 | ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this))); |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 86 | } |
| 87 | } |
| 88 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 89 | MediaSourceInterface::SourceState RemoteAudioSource::state() const { |
| 90 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 91 | return state_; |
| 92 | } |
| 93 | |
tommi | 6eca7e3 | 2015-12-15 04:27:11 -0800 | [diff] [blame] | 94 | bool RemoteAudioSource::remote() const { |
| 95 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 96 | return true; |
| 97 | } |
| 98 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 99 | void RemoteAudioSource::SetVolume(double volume) { |
| 100 | RTC_DCHECK(volume >= 0 && volume <= 10); |
| 101 | for (auto* observer : audio_observers_) |
| 102 | observer->OnSetVolume(volume); |
| 103 | } |
| 104 | |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 105 | void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 106 | RTC_DCHECK(observer != NULL); |
| 107 | RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(), |
| 108 | observer) == audio_observers_.end()); |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 109 | audio_observers_.push_back(observer); |
| 110 | } |
| 111 | |
| 112 | void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 113 | RTC_DCHECK(observer != NULL); |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 114 | audio_observers_.remove(observer); |
| 115 | } |
| 116 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 117 | void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) { |
| 118 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 119 | RTC_DCHECK(sink); |
| 120 | |
| 121 | if (state_ != MediaSourceInterface::kLive) { |
| 122 | LOG(LS_ERROR) << "Can't register sink as the source isn't live."; |
| 123 | return; |
| 124 | } |
| 125 | |
| 126 | rtc::CritScope lock(&sink_lock_); |
| 127 | RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end()); |
| 128 | sinks_.push_back(sink); |
| 129 | } |
| 130 | |
| 131 | void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) { |
| 132 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 133 | RTC_DCHECK(sink); |
| 134 | |
| 135 | rtc::CritScope lock(&sink_lock_); |
| 136 | sinks_.remove(sink); |
| 137 | } |
| 138 | |
| 139 | void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { |
| 140 | // Called on the externally-owned audio callback thread, via/from webrtc. |
| 141 | rtc::CritScope lock(&sink_lock_); |
| 142 | for (auto* sink : sinks_) { |
| 143 | sink->OnData(audio.data, 16, audio.sample_rate, audio.channels, |
| 144 | audio.samples_per_channel); |
| 145 | } |
| 146 | } |
| 147 | |
| 148 | void RemoteAudioSource::OnAudioProviderGone() { |
| 149 | // Called when the data provider is deleted. It may be the worker thread |
| 150 | // in libjingle or may be a different worker thread. |
Taylor Brandstetter | 5d97a9a | 2016-06-10 14:17:27 -0700 | [diff] [blame^] | 151 | main_thread_->Post(RTC_FROM_HERE, new MessageHandler(this)); |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 152 | } |
| 153 | |
| 154 | void RemoteAudioSource::OnMessage(rtc::Message* msg) { |
| 155 | RTC_DCHECK(main_thread_->IsCurrent()); |
| 156 | sinks_.clear(); |
| 157 | state_ = MediaSourceInterface::kEnded; |
| 158 | FireOnChanged(); |
| 159 | } |
| 160 | |
wu@webrtc.org | b9a088b | 2014-02-13 23:18:49 +0000 | [diff] [blame] | 161 | } // namespace webrtc |