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Side by Side Diff: webrtc/pc/channel.cc

Issue 1847353004: Allow applications to control audio send bitrate through RtpParameters. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: More CR feedback Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2004 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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1402 // our local variable. This is OK since we're synchronously invoking. 1402 // our local variable. This is OK since we're synchronously invoking.
1403 InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink)); 1403 InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
1404 } 1404 }
1405 1405
1406 webrtc::RtpParameters VoiceChannel::GetRtpParameters(uint32_t ssrc) const { 1406 webrtc::RtpParameters VoiceChannel::GetRtpParameters(uint32_t ssrc) const {
1407 return worker_thread()->Invoke<webrtc::RtpParameters>( 1407 return worker_thread()->Invoke<webrtc::RtpParameters>(
1408 Bind(&VoiceChannel::GetRtpParameters_w, this, ssrc)); 1408 Bind(&VoiceChannel::GetRtpParameters_w, this, ssrc));
1409 } 1409 }
1410 1410
1411 webrtc::RtpParameters VoiceChannel::GetRtpParameters_w(uint32_t ssrc) const { 1411 webrtc::RtpParameters VoiceChannel::GetRtpParameters_w(uint32_t ssrc) const {
1412 // Not yet implemented. 1412 return media_channel()->GetRtpParameters(ssrc);
1413 // TODO(skvlad): Add support for limiting send bitrate for audio channels.
1414 return webrtc::RtpParameters();
1415 } 1413 }
1416 1414
1417 bool VoiceChannel::SetRtpParameters(uint32_t ssrc, 1415 bool VoiceChannel::SetRtpParameters(uint32_t ssrc,
1418 const webrtc::RtpParameters& parameters) { 1416 const webrtc::RtpParameters& parameters) {
1419 return InvokeOnWorker( 1417 return InvokeOnWorker(
1420 Bind(&VoiceChannel::SetRtpParameters_w, this, ssrc, parameters)); 1418 Bind(&VoiceChannel::SetRtpParameters_w, this, ssrc, parameters));
1421 } 1419 }
1422 1420
1423 bool VoiceChannel::SetRtpParameters_w(uint32_t ssrc, 1421 bool VoiceChannel::SetRtpParameters_w(uint32_t ssrc,
1424 webrtc::RtpParameters parameters) { 1422 webrtc::RtpParameters parameters) {
1425 // Not yet implemented. 1423 return media_channel()->SetRtpParameters(ssrc, parameters);
1426 // TODO(skvlad): Add support for limiting send bitrate for audio channels.
1427 return false;
1428 } 1424 }
1429 1425
1430 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { 1426 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
1431 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats, 1427 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
1432 media_channel(), stats)); 1428 media_channel(), stats));
1433 } 1429 }
1434 1430
1435 void VoiceChannel::StartMediaMonitor(int cms) { 1431 void VoiceChannel::StartMediaMonitor(int cms) {
1436 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), 1432 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
1437 rtc::Thread::Current())); 1433 rtc::Thread::Current()));
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2205 return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp(); 2201 return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
2206 } 2202 }
2207 2203
2208 void DataChannel::OnStreamClosedRemotely(uint32_t sid) { 2204 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
2209 rtc::TypedMessageData<uint32_t>* message = 2205 rtc::TypedMessageData<uint32_t>* message =
2210 new rtc::TypedMessageData<uint32_t>(sid); 2206 new rtc::TypedMessageData<uint32_t>(sid);
2211 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message); 2207 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
2212 } 2208 }
2213 2209
2214 } // namespace cricket 2210 } // namespace cricket
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