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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, 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/rtpreceiver.h"
29
30 #include "talk/app/webrtc/videosourceinterface.h"
31
32 namespace webrtc {
33
34 AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
35 uint32 ssrc,
36 AudioProviderInterface* provider)
37 : track_(track),
38 ssrc_(ssrc),
39 provider_(provider),
40 cached_track_enabled_(track->enabled()) {
41 track_->RegisterObserver(this);
42 track_->GetSource()->RegisterAudioObserver(this);
43 UpdateProvider();
44 }
45
46 AudioRtpReceiver::~AudioRtpReceiver() {
47 track_->GetSource()->UnregisterAudioObserver(this);
48 track_->UnregisterObserver(this);
49 if (provider_) {
50 DetachFromProvider();
51 }
52 }
53
54 void AudioRtpReceiver::OnChanged() {
55 if (cached_track_enabled_ != track_->enabled()) {
56 cached_track_enabled_ = track_->enabled();
57 UpdateProvider();
58 }
59 }
60
61 void AudioRtpReceiver::OnSetVolume(double volume) {
62 // When the track is disabled, the volume of the source, which is the
63 // corresponding WebRtc Voice Engine channel will be 0. So we do not allow
64 // setting the volume to the source when the track is disabled.
65 if (provider_ && track_->enabled())
66 provider_->SetAudioPlayoutVolume(ssrc_, volume);
67 }
68
69 void AudioRtpReceiver::DetachFromProvider() {
70 provider_->SetAudioPlayout(ssrc_, false, nullptr);
pthatcher1 2015/09/23 14:47:39 If you put if (!provider_) { return; } At the
pthatcher1 2015/09/24 06:32:29 I still think this is a good idea.
71 provider_ = nullptr;
72 }
73
74 void AudioRtpReceiver::UpdateProvider() {
pthatcher1 2015/09/23 14:47:39 I think that it would make sense to call this some
pthatcher1 2015/09/24 06:32:29 I still think this is a good idea.
75 if (!provider_) {
76 return;
77 }
78 provider_->SetAudioPlayout(ssrc_, track_->enabled(), track_->GetRenderer());
79 }
80
81 VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track,
82 uint32 ssrc,
83 VideoProviderInterface* provider)
84 : track_(track), ssrc_(ssrc), provider_(provider) {
85 provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
86 }
87
88 VideoRtpReceiver::~VideoRtpReceiver() {
89 if (provider_) {
90 // Since cricket::VideoRenderer is not reference counted,
91 // we need to remove it from the provider before we are deleted.
92 DetachFromProvider();
93 }
94 }
95
96 void VideoRtpReceiver::DetachFromProvider() {
97 provider_->SetVideoPlayout(ssrc_, false, nullptr);
98 provider_ = nullptr;
99 }
100
101 } // namespace webrtc
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