blob: faf3de30338592f0e2a6f133ec42195d986eb6a2 [file] [log] [blame]
deadbeef6979b022015-09-24 16:47:53 -07001/*
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
deadbeef70ab1a12015-09-28 16:53:55 -070030#include "talk/app/webrtc/videosourceinterface.h"
31
32namespace webrtc {
33
34AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
Peter Boström0c4e06b2015-10-07 12:23:21 +020035 uint32_t ssrc,
deadbeef70ab1a12015-09-28 16:53:55 -070036 AudioProviderInterface* provider)
37 : id_(track->id()),
38 track_(track),
39 ssrc_(ssrc),
40 provider_(provider),
41 cached_track_enabled_(track->enabled()) {
42 track_->RegisterObserver(this);
43 track_->GetSource()->RegisterAudioObserver(this);
44 Reconfigure();
45}
46
47AudioRtpReceiver::~AudioRtpReceiver() {
48 track_->GetSource()->UnregisterAudioObserver(this);
49 track_->UnregisterObserver(this);
50 Stop();
51}
52
53void AudioRtpReceiver::OnChanged() {
54 if (cached_track_enabled_ != track_->enabled()) {
55 cached_track_enabled_ = track_->enabled();
56 Reconfigure();
57 }
58}
59
60void AudioRtpReceiver::OnSetVolume(double volume) {
61 // When the track is disabled, the volume of the source, which is the
62 // corresponding WebRtc Voice Engine channel will be 0. So we do not allow
63 // setting the volume to the source when the track is disabled.
64 if (provider_ && track_->enabled())
65 provider_->SetAudioPlayoutVolume(ssrc_, volume);
66}
67
68void AudioRtpReceiver::Stop() {
69 // TODO(deadbeef): Need to do more here to fully stop receiving packets.
70 if (!provider_) {
71 return;
72 }
73 provider_->SetAudioPlayout(ssrc_, false, nullptr);
74 provider_ = nullptr;
75}
76
77void AudioRtpReceiver::Reconfigure() {
78 if (!provider_) {
79 return;
80 }
81 provider_->SetAudioPlayout(ssrc_, track_->enabled(), track_->GetRenderer());
82}
83
84VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track,
Peter Boström0c4e06b2015-10-07 12:23:21 +020085 uint32_t ssrc,
deadbeef70ab1a12015-09-28 16:53:55 -070086 VideoProviderInterface* provider)
87 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
88 provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
89}
90
91VideoRtpReceiver::~VideoRtpReceiver() {
92 // Since cricket::VideoRenderer is not reference counted,
93 // we need to remove it from the provider before we are deleted.
94 Stop();
95}
96
97void VideoRtpReceiver::Stop() {
98 // TODO(deadbeef): Need to do more here to fully stop receiving packets.
99 if (!provider_) {
100 return;
101 }
102 provider_->SetVideoPlayout(ssrc_, false, nullptr);
103 provider_ = nullptr;
104}
105
106} // namespace webrtc