Chromium Code Reviews| OLD | NEW |
|---|---|
| 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 |
| (...skipping 1645 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 1656 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, | 1656 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc, |
| 1657 webrtc::RtpParameters parameters) { | 1657 webrtc::RtpParameters parameters) { |
| 1658 return media_channel()->SetRtpReceiveParameters(ssrc, parameters); | 1658 return media_channel()->SetRtpReceiveParameters(ssrc, parameters); |
| 1659 } | 1659 } |
| 1660 | 1660 |
| 1661 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { | 1661 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) { |
| 1662 return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, | 1662 return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats, |
| 1663 media_channel(), stats)); | 1663 media_channel(), stats)); |
| 1664 } | 1664 } |
| 1665 | 1665 |
| 1666 std::vector<webrtc::RtpContributingSource*> | |
| 1667 VoiceChannel::GetContributingSources(uint32_t ssrc) { | |
| 1668 return worker_thread()->Invoke<std::vector<webrtc::RtpContributingSource*>>( | |
| 1669 RTC_FROM_HERE, | |
| 1670 Bind(&VoiceMediaChannel::GetContributingSources, media_channel(), ssrc)); | |
|
Zhi Huang
2017/03/31 06:44:04
I'll downcast this instead of adding the method to
| |
| 1671 } | |
| 1672 | |
| 1666 void VoiceChannel::StartMediaMonitor(int cms) { | 1673 void VoiceChannel::StartMediaMonitor(int cms) { |
| 1667 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), | 1674 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(), |
| 1668 rtc::Thread::Current())); | 1675 rtc::Thread::Current())); |
| 1669 media_monitor_->SignalUpdate.connect( | 1676 media_monitor_->SignalUpdate.connect( |
| 1670 this, &VoiceChannel::OnMediaMonitorUpdate); | 1677 this, &VoiceChannel::OnMediaMonitorUpdate); |
| 1671 media_monitor_->Start(cms); | 1678 media_monitor_->Start(cms); |
| 1672 } | 1679 } |
| 1673 | 1680 |
| 1674 void VoiceChannel::StopMediaMonitor() { | 1681 void VoiceChannel::StopMediaMonitor() { |
| 1675 if (media_monitor_) { | 1682 if (media_monitor_) { |
| (...skipping 789 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 2465 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, | 2472 signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA, |
| 2466 new DataChannelReadyToSendMessageData(writable)); | 2473 new DataChannelReadyToSendMessageData(writable)); |
| 2467 } | 2474 } |
| 2468 | 2475 |
| 2469 void RtpDataChannel::GetSrtpCryptoSuites_n( | 2476 void RtpDataChannel::GetSrtpCryptoSuites_n( |
| 2470 std::vector<int>* crypto_suites) const { | 2477 std::vector<int>* crypto_suites) const { |
| 2471 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); | 2478 GetSupportedDataCryptoSuites(crypto_options(), crypto_suites); |
| 2472 } | 2479 } |
| 2473 | 2480 |
| 2474 } // namespace cricket | 2481 } // namespace cricket |
| OLD | NEW |