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

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing indentation. Created 5 years, 2 months ago
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1 /*
2 * libjingle
3 * Copyright 2015 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 #include "talk/app/webrtc/rtpsender.h"
29
30 #include "talk/app/webrtc/localaudiosource.h"
31 #include "talk/app/webrtc/videosourceinterface.h"
32
33 namespace webrtc {
34
35 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
36
37 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
38 rtc::CritScope lock(&lock_);
39 if (sink_)
40 sink_->OnClose();
41 }
42
43 void LocalAudioSinkAdapter::OnData(const void* audio_data,
44 int bits_per_sample,
45 int sample_rate,
46 int number_of_channels,
47 size_t number_of_frames) {
48 rtc::CritScope lock(&lock_);
49 if (sink_) {
50 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
51 number_of_frames);
52 }
53 }
54
55 void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
56 rtc::CritScope lock(&lock_);
57 ASSERT(!sink || !sink_);
58 sink_ = sink;
59 }
60
61 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
62 uint32 ssrc,
63 AudioProviderInterface* provider)
64 : track_(track),
65 ssrc_(ssrc),
66 provider_(provider),
67 cached_track_enabled_(track->enabled()),
68 sink_adapter_(new LocalAudioSinkAdapter()) {
69 track_->RegisterObserver(this);
70 track_->AddSink(sink_adapter_.get());
71 UpdateProvider();
72 }
73
74 AudioRtpSender::~AudioRtpSender() {
75 track_->RemoveSink(sink_adapter_.get());
76 track_->UnregisterObserver(this);
77 if (provider_) {
78 DetachFromProvider();
79 }
80 }
81
82 void AudioRtpSender::OnChanged() {
83 if (cached_track_enabled_ != track_->enabled()) {
84 cached_track_enabled_ = track_->enabled();
85 UpdateProvider();
86 }
87 }
88
89 void AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
90 if (track->kind() != "audio") {
91 LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
92 << " track.";
93 return;
94 }
95 AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
96
97 // Detach from old track.
98 track_->RemoveSink(sink_adapter_.get());
99 track_->UnregisterObserver(this);
100
101 // Attach to new track.
102 track_ = audio_track;
103 cached_track_enabled_ = track_->enabled();
104 track_->RegisterObserver(this);
105 track_->AddSink(sink_adapter_.get());
106 UpdateProvider();
107 }
108
109 void AudioRtpSender::DetachFromProvider() {
110 cricket::AudioOptions options;
111 provider_->SetAudioSend(ssrc_, false, options, nullptr);
112 provider_ = nullptr;
113 }
114
115 void AudioRtpSender::UpdateProvider() {
116 if (!provider_) {
117 return;
118 }
119 cricket::AudioOptions options;
120 if (track_->enabled() && track_->GetSource()) {
121 // TODO(xians): Remove this static_cast since we should be able to connect
122 // a remote audio track to peer connection.
123 options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
124 }
125
126 // Use the renderer if the audio track has one, otherwise use the sink
127 // adapter owned by this class.
128 cricket::AudioRenderer* renderer =
129 track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
130 ASSERT(renderer != nullptr);
131 provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
132 }
133
134 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
135 uint32 ssrc,
136 VideoProviderInterface* provider)
137 : track_(track),
138 ssrc_(ssrc),
139 provider_(provider),
140 cached_track_enabled_(track->enabled()) {
141 track_->RegisterObserver(this);
142 VideoSourceInterface* source = track_->GetSource();
143 if (source) {
144 provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
145 }
146 UpdateProvider();
147 }
148
149 VideoRtpSender::~VideoRtpSender() {
150 track_->UnregisterObserver(this);
151 if (provider_) {
152 DetachFromProvider();
153 }
154 }
155
156 void VideoRtpSender::OnChanged() {
157 if (cached_track_enabled_ != track_->enabled()) {
158 cached_track_enabled_ = track_->enabled();
159 UpdateProvider();
160 }
161 }
162
163 void VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
164 if (track->kind() != "video") {
165 LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
166 << " track.";
167 return;
168 }
169 VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
170
171 // Detach from old track.
172 track_->UnregisterObserver(this);
173
174 // Attach to new track.
175 track_ = video_track;
176 cached_track_enabled_ = track_->enabled();
177 track_->RegisterObserver(this);
178 UpdateProvider();
179 }
180
181 void VideoRtpSender::DetachFromProvider() {
182 provider_->SetCaptureDevice(ssrc_, nullptr);
183 provider_->SetVideoSend(ssrc_, false, nullptr);
184 provider_ = nullptr;
185 }
186
187 void VideoRtpSender::UpdateProvider() {
188 if (!provider_) {
189 return;
190 }
191 const cricket::VideoOptions* options = nullptr;
192 VideoSourceInterface* source = track_->GetSource();
193 if (track_->enabled() && source) {
194 options = source->options();
195 }
196 provider_->SetVideoSend(ssrc_, track_->enabled(), options);
197 }
198
199 } // namespace webrtc
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