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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: Created 5 years, 3 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_(NULL) {
36 }
37
38 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
39 rtc::CritScope lock(&lock_);
40 if (sink_)
41 sink_->OnClose();
42 }
43
44 void LocalAudioSinkAdapter::OnData(const void* audio_data,
45 int bits_per_sample,
46 int sample_rate,
47 int number_of_channels,
48 size_t number_of_frames) {
49 rtc::CritScope lock(&lock_);
50 if (sink_) {
51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
52 number_of_frames);
53 }
54 }
55
56 void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
57 rtc::CritScope lock(&lock_);
58 ASSERT(!sink || !sink_);
59 sink_ = sink;
60 }
61
62 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
63 uint32 ssrc,
64 const std::string& mid,
65 AudioProviderInterface* provider)
66 : track_(track),
67 ssrc_(ssrc),
68 mid_(mid),
69 provider_(provider),
70 enabled_(track->enabled()),
71 sink_adapter_(new LocalAudioSinkAdapter()) {
72 UpdateEnabled();
73 track_->RegisterObserver(this);
74 track_->AddSink(sink_adapter_.get());
75 }
76
77 AudioRtpSender::~AudioRtpSender() {
78 track_->RemoveSink(sink_adapter_.get());
79 track_->UnregisterObserver(this);
80 if (provider_) {
81 Stop();
82 }
83 }
84
85 void AudioRtpSender::OnChanged() {
86 if (enabled_ != track_->enabled()) {
87 enabled_ = track_->enabled();
88 UpdateEnabled();
89 }
90 }
91
92 void AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
93 if (track->kind() != "audio") {
94 LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
95 << " track.";
96 return;
97 }
98 AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
99 track_->RemoveSink(sink_adapter_.get());
100 track_->UnregisterObserver(this);
101 track_ = audio_track;
102 track_->RegisterObserver(this);
103 track_->AddSink(sink_adapter_.get());
104 // Check if the new track has a different "enabled" setting
105 OnChanged();
106 }
107
108 void AudioRtpSender::Stop() {
109 cricket::AudioOptions options;
110 provider_->SetAudioSend(ssrc_, false, options, nullptr);
111 provider_ = nullptr;
112 }
113
114 void AudioRtpSender::UpdateEnabled() {
115 if (!provider_) {
116 return;
117 }
118 cricket::AudioOptions options;
119 if (track_->enabled() && track_->GetSource()) {
120 // TODO(xians): Remove this static_cast since we should be able to connect
121 // a remote audio track to peer connection.
122 options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
123 }
124
125 // Use the renderer if the audio track has one, otherwise use the sink
126 // adapter owned by this class.
127 cricket::AudioRenderer* renderer =
128 track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
129 ASSERT(renderer != nullptr);
130 provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
131 }
132
133 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
134 uint32 ssrc,
135 const std::string& mid,
136 VideoProviderInterface* provider)
137 : track_(track),
138 ssrc_(ssrc),
139 mid_(mid),
140 provider_(provider),
141 enabled_(track->enabled()) {
142 track_->RegisterObserver(this);
143 VideoSourceInterface* source = track_->GetSource();
144 if (source)
145 provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
146 UpdateEnabled();
147 }
148
149 VideoRtpSender::~VideoRtpSender() {
150 track_->UnregisterObserver(this);
151 if (provider_) {
152 Stop();
153 }
154 }
155
156 void VideoRtpSender::OnChanged() {
157 if (enabled_ != track_->enabled()) {
158 enabled_ = track_->enabled();
159 UpdateEnabled();
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 track_->UnregisterObserver(this);
171 track_ = video_track;
172 track_->RegisterObserver(this);
173 // Check if the new track has a different "enabled" setting.
174 OnChanged();
175 }
176
177 void VideoRtpSender::Stop() {
178 provider_->SetCaptureDevice(ssrc_, nullptr);
179 provider_->SetVideoSend(ssrc_, false, nullptr);
180 provider_ = nullptr;
181 }
182
183 void VideoRtpSender::UpdateEnabled() {
184 if (!provider_) {
185 return;
186 }
187 const cricket::VideoOptions* options = nullptr;
188 VideoSourceInterface* source = track_->GetSource();
189 if (track_->enabled() && source) {
190 options = source->options();
191 }
192 provider_->SetVideoSend(ssrc_, track_->enabled(), options);
193 }
194
195 } // namespace webrtc
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