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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_t ssrc, | |
| 36 AudioProviderInterface* provider) | |
| 37 : id_(track->id()), | |
| 38 track_(track), | |
| 39 ssrc_(ssrc), | |
| 40 provider_(provider), | |
| 41 cached_track_enabled_(track->enabled()) { | |
| 42 RTC_DCHECK(track_->GetSource()->remote()); | |
| 43 track_->RegisterObserver(this); | |
| 44 track_->GetSource()->RegisterAudioObserver(this); | |
| 45 Reconfigure(); | |
| 46 } | |
| 47 | |
| 48 AudioRtpReceiver::~AudioRtpReceiver() { | |
| 49 track_->GetSource()->UnregisterAudioObserver(this); | |
| 50 track_->UnregisterObserver(this); | |
| 51 Stop(); | |
| 52 } | |
| 53 | |
| 54 void AudioRtpReceiver::OnChanged() { | |
| 55 if (cached_track_enabled_ != track_->enabled()) { | |
| 56 cached_track_enabled_ = track_->enabled(); | |
| 57 Reconfigure(); | |
| 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::Stop() { | |
| 70 // TODO(deadbeef): Need to do more here to fully stop receiving packets. | |
| 71 if (!provider_) { | |
| 72 return; | |
| 73 } | |
| 74 provider_->SetAudioPlayout(ssrc_, false); | |
| 75 provider_ = nullptr; | |
| 76 } | |
| 77 | |
| 78 void AudioRtpReceiver::Reconfigure() { | |
| 79 if (!provider_) { | |
| 80 return; | |
| 81 } | |
| 82 provider_->SetAudioPlayout(ssrc_, track_->enabled()); | |
| 83 } | |
| 84 | |
| 85 VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track, | |
| 86 uint32_t ssrc, | |
| 87 VideoProviderInterface* provider) | |
| 88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) { | |
| 89 provider_->SetVideoPlayout(ssrc_, true, track_->GetSink()); | |
| 90 } | |
| 91 | |
| 92 VideoRtpReceiver::~VideoRtpReceiver() { | |
| 93 // Since cricket::VideoRenderer is not reference counted, | |
| 94 // we need to remove it from the provider before we are deleted. | |
| 95 Stop(); | |
| 96 } | |
| 97 | |
| 98 void VideoRtpReceiver::Stop() { | |
| 99 // TODO(deadbeef): Need to do more here to fully stop receiving packets. | |
| 100 if (!provider_) { | |
| 101 return; | |
| 102 } | |
| 103 provider_->SetVideoPlayout(ssrc_, false, nullptr); | |
| 104 provider_ = nullptr; | |
| 105 } | |
| 106 | |
| 107 } // namespace webrtc | |
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Chromium Code Reviews
Issue 1610243002:
Move talk/app/webrtc to webrtc/api (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master