deadbeef | 6979b02 | 2015-09-24 16:47:53 -0700 | [diff] [blame] | 1 | /* |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 2 | * Copyright 2015 The WebRTC project authors. All Rights Reserved. |
deadbeef | 6979b02 | 2015-09-24 16:47:53 -0700 | [diff] [blame] | 3 | * |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
deadbeef | 6979b02 | 2015-09-24 16:47:53 -0700 | [diff] [blame] | 9 | */ |
| 10 | |
Henrik Kjellander | 15583c1 | 2016-02-10 10:53:12 +0100 | [diff] [blame] | 11 | #include "webrtc/api/rtpsender.h" |
deadbeef | 6979b02 | 2015-09-24 16:47:53 -0700 | [diff] [blame] | 12 | |
Henrik Kjellander | 15583c1 | 2016-02-10 10:53:12 +0100 | [diff] [blame] | 13 | #include "webrtc/api/localaudiosource.h" |
perkj | 9e083d2 | 2016-03-20 09:38:40 -0700 | [diff] [blame] | 14 | #include "webrtc/api/mediastreaminterface.h" |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 15 | #include "webrtc/base/helpers.h" |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 16 | #include "webrtc/base/trace_event.h" |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 17 | |
| 18 | namespace webrtc { |
| 19 | |
| 20 | LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {} |
| 21 | |
| 22 | LocalAudioSinkAdapter::~LocalAudioSinkAdapter() { |
| 23 | rtc::CritScope lock(&lock_); |
| 24 | if (sink_) |
| 25 | sink_->OnClose(); |
| 26 | } |
| 27 | |
| 28 | void LocalAudioSinkAdapter::OnData(const void* audio_data, |
| 29 | int bits_per_sample, |
| 30 | int sample_rate, |
Peter Kasting | 6955870 | 2016-01-12 16:26:35 -0800 | [diff] [blame] | 31 | size_t number_of_channels, |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 32 | size_t number_of_frames) { |
| 33 | rtc::CritScope lock(&lock_); |
| 34 | if (sink_) { |
| 35 | sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, |
| 36 | number_of_frames); |
| 37 | } |
| 38 | } |
| 39 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 40 | void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 41 | rtc::CritScope lock(&lock_); |
| 42 | ASSERT(!sink || !sink_); |
| 43 | sink_ = sink; |
| 44 | } |
| 45 | |
| 46 | AudioRtpSender::AudioRtpSender(AudioTrackInterface* track, |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 47 | const std::string& stream_id, |
| 48 | AudioProviderInterface* provider, |
| 49 | StatsCollector* stats) |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 50 | : id_(track->id()), |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 51 | stream_id_(stream_id), |
deadbeef | 5def7b9 | 2015-11-20 11:43:22 -0800 | [diff] [blame] | 52 | provider_(provider), |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 53 | stats_(stats), |
| 54 | track_(track), |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 55 | cached_track_enabled_(track->enabled()), |
| 56 | sink_adapter_(new LocalAudioSinkAdapter()) { |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 57 | RTC_DCHECK(provider != nullptr); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 58 | track_->RegisterObserver(this); |
| 59 | track_->AddSink(sink_adapter_.get()); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 60 | } |
| 61 | |
deadbeef | e1f9d83 | 2016-01-14 15:35:42 -0800 | [diff] [blame] | 62 | AudioRtpSender::AudioRtpSender(AudioTrackInterface* track, |
| 63 | AudioProviderInterface* provider, |
| 64 | StatsCollector* stats) |
| 65 | : id_(track->id()), |
| 66 | stream_id_(rtc::CreateRandomUuid()), |
| 67 | provider_(provider), |
| 68 | stats_(stats), |
| 69 | track_(track), |
| 70 | cached_track_enabled_(track->enabled()), |
| 71 | sink_adapter_(new LocalAudioSinkAdapter()) { |
| 72 | RTC_DCHECK(provider != nullptr); |
| 73 | track_->RegisterObserver(this); |
| 74 | track_->AddSink(sink_adapter_.get()); |
| 75 | } |
| 76 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 77 | AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider, |
| 78 | StatsCollector* stats) |
| 79 | : id_(rtc::CreateRandomUuid()), |
| 80 | stream_id_(rtc::CreateRandomUuid()), |
| 81 | provider_(provider), |
| 82 | stats_(stats), |
| 83 | sink_adapter_(new LocalAudioSinkAdapter()) {} |
| 84 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 85 | AudioRtpSender::~AudioRtpSender() { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 86 | Stop(); |
| 87 | } |
| 88 | |
| 89 | void AudioRtpSender::OnChanged() { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 90 | TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 91 | RTC_DCHECK(!stopped_); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 92 | if (cached_track_enabled_ != track_->enabled()) { |
| 93 | cached_track_enabled_ = track_->enabled(); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 94 | if (can_send_track()) { |
| 95 | SetAudioSend(); |
| 96 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 97 | } |
| 98 | } |
| 99 | |
| 100 | bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 101 | TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 102 | if (stopped_) { |
| 103 | LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender."; |
| 104 | return false; |
| 105 | } |
| 106 | if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 107 | LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind() |
| 108 | << " track."; |
| 109 | return false; |
| 110 | } |
| 111 | AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track); |
| 112 | |
| 113 | // Detach from old track. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 114 | if (track_) { |
| 115 | track_->RemoveSink(sink_adapter_.get()); |
| 116 | track_->UnregisterObserver(this); |
| 117 | } |
| 118 | |
| 119 | if (can_send_track() && stats_) { |
| 120 | stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); |
| 121 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 122 | |
| 123 | // Attach to new track. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 124 | bool prev_can_send_track = can_send_track(); |
deadbeef | 5dd42fd | 2016-05-02 16:20:01 -0700 | [diff] [blame] | 125 | // Keep a reference to the old track to keep it alive until we call |
| 126 | // SetAudioSend. |
| 127 | rtc::scoped_refptr<AudioTrackInterface> old_track = track_; |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 128 | track_ = audio_track; |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 129 | if (track_) { |
| 130 | cached_track_enabled_ = track_->enabled(); |
| 131 | track_->RegisterObserver(this); |
| 132 | track_->AddSink(sink_adapter_.get()); |
| 133 | } |
| 134 | |
| 135 | // Update audio provider. |
| 136 | if (can_send_track()) { |
| 137 | SetAudioSend(); |
| 138 | if (stats_) { |
| 139 | stats_->AddLocalAudioTrack(track_.get(), ssrc_); |
| 140 | } |
| 141 | } else if (prev_can_send_track) { |
| 142 | cricket::AudioOptions options; |
| 143 | provider_->SetAudioSend(ssrc_, false, options, nullptr); |
| 144 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 145 | return true; |
| 146 | } |
| 147 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 148 | void AudioRtpSender::SetSsrc(uint32_t ssrc) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 149 | TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 150 | if (stopped_ || ssrc == ssrc_) { |
| 151 | return; |
| 152 | } |
| 153 | // If we are already sending with a particular SSRC, stop sending. |
| 154 | if (can_send_track()) { |
| 155 | cricket::AudioOptions options; |
| 156 | provider_->SetAudioSend(ssrc_, false, options, nullptr); |
| 157 | if (stats_) { |
| 158 | stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); |
| 159 | } |
| 160 | } |
| 161 | ssrc_ = ssrc; |
| 162 | if (can_send_track()) { |
| 163 | SetAudioSend(); |
| 164 | if (stats_) { |
| 165 | stats_->AddLocalAudioTrack(track_.get(), ssrc_); |
| 166 | } |
| 167 | } |
| 168 | } |
| 169 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 170 | void AudioRtpSender::Stop() { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 171 | TRACE_EVENT0("webrtc", "AudioRtpSender::Stop"); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 172 | // TODO(deadbeef): Need to do more here to fully stop sending packets. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 173 | if (stopped_) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 174 | return; |
| 175 | } |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 176 | if (track_) { |
| 177 | track_->RemoveSink(sink_adapter_.get()); |
| 178 | track_->UnregisterObserver(this); |
| 179 | } |
| 180 | if (can_send_track()) { |
| 181 | cricket::AudioOptions options; |
| 182 | provider_->SetAudioSend(ssrc_, false, options, nullptr); |
| 183 | if (stats_) { |
| 184 | stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); |
| 185 | } |
| 186 | } |
| 187 | stopped_ = true; |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 188 | } |
| 189 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 190 | void AudioRtpSender::SetAudioSend() { |
| 191 | RTC_DCHECK(!stopped_ && can_send_track()); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 192 | cricket::AudioOptions options; |
Tommi | 3c16978 | 2016-01-21 16:12:17 +0100 | [diff] [blame] | 193 | #if !defined(WEBRTC_CHROMIUM_BUILD) |
| 194 | // TODO(tommi): Remove this hack when we move CreateAudioSource out of |
| 195 | // PeerConnection. This is a bit of a strange way to apply local audio |
| 196 | // options since it is also applied to all streams/channels, local or remote. |
tommi | 6eca7e3 | 2015-12-15 04:27:11 -0800 | [diff] [blame] | 197 | if (track_->enabled() && track_->GetSource() && |
| 198 | !track_->GetSource()->remote()) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 199 | // TODO(xians): Remove this static_cast since we should be able to connect |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 200 | // a remote audio track to a peer connection. |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 201 | options = static_cast<LocalAudioSource*>(track_->GetSource())->options(); |
| 202 | } |
Tommi | 3c16978 | 2016-01-21 16:12:17 +0100 | [diff] [blame] | 203 | #endif |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 204 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 205 | cricket::AudioSource* source = sink_adapter_.get(); |
| 206 | ASSERT(source != nullptr); |
| 207 | provider_->SetAudioSend(ssrc_, track_->enabled(), options, source); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 208 | } |
| 209 | |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 210 | RtpParameters AudioRtpSender::GetParameters() const { |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 211 | return provider_->GetAudioRtpSendParameters(ssrc_); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 212 | } |
| 213 | |
| 214 | bool AudioRtpSender::SetParameters(const RtpParameters& parameters) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 215 | TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters"); |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 216 | return provider_->SetAudioRtpSendParameters(ssrc_, parameters); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 217 | } |
| 218 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 219 | VideoRtpSender::VideoRtpSender(VideoTrackInterface* track, |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 220 | const std::string& stream_id, |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 221 | VideoProviderInterface* provider) |
| 222 | : id_(track->id()), |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 223 | stream_id_(stream_id), |
deadbeef | 5def7b9 | 2015-11-20 11:43:22 -0800 | [diff] [blame] | 224 | provider_(provider), |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 225 | track_(track), |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 226 | cached_track_enabled_(track->enabled()) { |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 227 | RTC_DCHECK(provider != nullptr); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 228 | track_->RegisterObserver(this); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 229 | } |
| 230 | |
deadbeef | e1f9d83 | 2016-01-14 15:35:42 -0800 | [diff] [blame] | 231 | VideoRtpSender::VideoRtpSender(VideoTrackInterface* track, |
| 232 | VideoProviderInterface* provider) |
| 233 | : id_(track->id()), |
| 234 | stream_id_(rtc::CreateRandomUuid()), |
| 235 | provider_(provider), |
| 236 | track_(track), |
| 237 | cached_track_enabled_(track->enabled()) { |
| 238 | RTC_DCHECK(provider != nullptr); |
| 239 | track_->RegisterObserver(this); |
| 240 | } |
| 241 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 242 | VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider) |
| 243 | : id_(rtc::CreateRandomUuid()), |
| 244 | stream_id_(rtc::CreateRandomUuid()), |
| 245 | provider_(provider) {} |
| 246 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 247 | VideoRtpSender::~VideoRtpSender() { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 248 | Stop(); |
| 249 | } |
| 250 | |
| 251 | void VideoRtpSender::OnChanged() { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 252 | TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 253 | RTC_DCHECK(!stopped_); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 254 | if (cached_track_enabled_ != track_->enabled()) { |
| 255 | cached_track_enabled_ = track_->enabled(); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 256 | if (can_send_track()) { |
| 257 | SetVideoSend(); |
| 258 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 259 | } |
| 260 | } |
| 261 | |
| 262 | bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 263 | TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 264 | if (stopped_) { |
| 265 | LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender."; |
| 266 | return false; |
| 267 | } |
| 268 | if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 269 | LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind() |
| 270 | << " track."; |
| 271 | return false; |
| 272 | } |
| 273 | VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track); |
| 274 | |
| 275 | // Detach from old track. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 276 | if (track_) { |
| 277 | track_->UnregisterObserver(this); |
| 278 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 279 | |
| 280 | // Attach to new track. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 281 | bool prev_can_send_track = can_send_track(); |
deadbeef | 5dd42fd | 2016-05-02 16:20:01 -0700 | [diff] [blame] | 282 | // Keep a reference to the old track to keep it alive until we call |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 283 | // SetVideoSend. |
deadbeef | 5dd42fd | 2016-05-02 16:20:01 -0700 | [diff] [blame] | 284 | rtc::scoped_refptr<VideoTrackInterface> old_track = track_; |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 285 | track_ = video_track; |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 286 | if (track_) { |
| 287 | cached_track_enabled_ = track_->enabled(); |
| 288 | track_->RegisterObserver(this); |
| 289 | } |
| 290 | |
| 291 | // Update video provider. |
| 292 | if (can_send_track()) { |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 293 | SetVideoSend(); |
| 294 | } else if (prev_can_send_track) { |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 295 | ClearVideoSend(); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 296 | } |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 297 | return true; |
| 298 | } |
| 299 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 300 | void VideoRtpSender::SetSsrc(uint32_t ssrc) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 301 | TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc"); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 302 | if (stopped_ || ssrc == ssrc_) { |
| 303 | return; |
| 304 | } |
| 305 | // If we are already sending with a particular SSRC, stop sending. |
| 306 | if (can_send_track()) { |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 307 | ClearVideoSend(); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 308 | } |
| 309 | ssrc_ = ssrc; |
| 310 | if (can_send_track()) { |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 311 | SetVideoSend(); |
| 312 | } |
| 313 | } |
| 314 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 315 | void VideoRtpSender::Stop() { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 316 | TRACE_EVENT0("webrtc", "VideoRtpSender::Stop"); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 317 | // TODO(deadbeef): Need to do more here to fully stop sending packets. |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 318 | if (stopped_) { |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 319 | return; |
| 320 | } |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 321 | if (track_) { |
| 322 | track_->UnregisterObserver(this); |
| 323 | } |
| 324 | if (can_send_track()) { |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 325 | ClearVideoSend(); |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 326 | } |
| 327 | stopped_ = true; |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 328 | } |
| 329 | |
deadbeef | fac0655 | 2015-11-25 11:26:01 -0800 | [diff] [blame] | 330 | void VideoRtpSender::SetVideoSend() { |
| 331 | RTC_DCHECK(!stopped_ && can_send_track()); |
perkj | 0d3eef2 | 2016-03-09 02:39:17 +0100 | [diff] [blame] | 332 | cricket::VideoOptions options; |
perkj | a3ede6c | 2016-03-08 01:27:48 +0100 | [diff] [blame] | 333 | VideoTrackSourceInterface* source = track_->GetSource(); |
perkj | 0d3eef2 | 2016-03-09 02:39:17 +0100 | [diff] [blame] | 334 | if (source) { |
| 335 | options.is_screencast = rtc::Optional<bool>(source->is_screencast()); |
Per | c0d31e9 | 2016-03-31 17:23:39 +0200 | [diff] [blame] | 336 | options.video_noise_reduction = source->needs_denoising(); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 337 | } |
deadbeef | 5a4a75a | 2016-06-02 16:23:38 -0700 | [diff] [blame] | 338 | provider_->SetVideoSend(ssrc_, track_->enabled(), &options, track_); |
| 339 | } |
| 340 | |
| 341 | void VideoRtpSender::ClearVideoSend() { |
| 342 | RTC_DCHECK(ssrc_ != 0); |
| 343 | RTC_DCHECK(provider_ != nullptr); |
| 344 | provider_->SetVideoSend(ssrc_, false, nullptr, nullptr); |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 345 | } |
| 346 | |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 347 | RtpParameters VideoRtpSender::GetParameters() const { |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 348 | return provider_->GetVideoRtpSendParameters(ssrc_); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 349 | } |
| 350 | |
| 351 | bool VideoRtpSender::SetParameters(const RtpParameters& parameters) { |
Peter Boström | dabc944 | 2016-04-11 11:45:14 +0200 | [diff] [blame] | 352 | TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters"); |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 353 | return provider_->SetVideoRtpSendParameters(ssrc_, parameters); |
skvlad | dc1c62c | 2016-03-16 19:07:43 -0700 | [diff] [blame] | 354 | } |
| 355 | |
deadbeef | 70ab1a1 | 2015-09-28 16:53:55 -0700 | [diff] [blame] | 356 | } // namespace webrtc |