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Issue 1917193008: Adding getParameters/setParameters APIs to RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: objc compile errors Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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201 options = static_cast<LocalAudioSource*>(track_->GetSource())->options(); 201 options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
202 } 202 }
203 #endif 203 #endif
204 204
205 cricket::AudioSource* source = sink_adapter_.get(); 205 cricket::AudioSource* source = sink_adapter_.get();
206 ASSERT(source != nullptr); 206 ASSERT(source != nullptr);
207 provider_->SetAudioSend(ssrc_, track_->enabled(), options, source); 207 provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
208 } 208 }
209 209
210 RtpParameters AudioRtpSender::GetParameters() const { 210 RtpParameters AudioRtpSender::GetParameters() const {
211 return provider_->GetAudioRtpParameters(ssrc_); 211 return provider_->GetAudioRtpSendParameters(ssrc_);
212 } 212 }
213 213
214 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) { 214 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
215 TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters"); 215 TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
216 return provider_->SetAudioRtpParameters(ssrc_, parameters); 216 return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
217 } 217 }
218 218
219 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track, 219 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
220 const std::string& stream_id, 220 const std::string& stream_id,
221 VideoProviderInterface* provider) 221 VideoProviderInterface* provider)
222 : id_(track->id()), 222 : id_(track->id()),
223 stream_id_(stream_id), 223 stream_id_(stream_id),
224 provider_(provider), 224 provider_(provider),
225 track_(track), 225 track_(track),
226 cached_track_enabled_(track->enabled()) { 226 cached_track_enabled_(track->enabled()) {
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343 cricket::VideoOptions options; 343 cricket::VideoOptions options;
344 VideoTrackSourceInterface* source = track_->GetSource(); 344 VideoTrackSourceInterface* source = track_->GetSource();
345 if (source) { 345 if (source) {
346 options.is_screencast = rtc::Optional<bool>(source->is_screencast()); 346 options.is_screencast = rtc::Optional<bool>(source->is_screencast());
347 options.video_noise_reduction = source->needs_denoising(); 347 options.video_noise_reduction = source->needs_denoising();
348 } 348 }
349 provider_->SetVideoSend(ssrc_, track_->enabled(), &options); 349 provider_->SetVideoSend(ssrc_, track_->enabled(), &options);
350 } 350 }
351 351
352 RtpParameters VideoRtpSender::GetParameters() const { 352 RtpParameters VideoRtpSender::GetParameters() const {
353 return provider_->GetVideoRtpParameters(ssrc_); 353 return provider_->GetVideoRtpSendParameters(ssrc_);
354 } 354 }
355 355
356 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) { 356 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
357 TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters"); 357 TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
358 return provider_->SetVideoRtpParameters(ssrc_, parameters); 358 return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
359 } 359 }
360 360
361 } // namespace webrtc 361 } // namespace webrtc
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