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| 1 /* | 1 /* |
| 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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| 2327 | 2327 |
| 2328 // SR may continue RR and any RR entry may correspond to any one of the send | 2328 // SR may continue RR and any RR entry may correspond to any one of the send |
| 2329 // channels. So all RTCP packets must be forwarded all send channels. VoE | 2329 // channels. So all RTCP packets must be forwarded all send channels. VoE |
| 2330 // will filter out RR internally. | 2330 // will filter out RR internally. |
| 2331 for (const auto& ch : send_streams_) { | 2331 for (const auto& ch : send_streams_) { |
| 2332 engine()->voe()->network()->ReceivedRTCPPacket( | 2332 engine()->voe()->network()->ReceivedRTCPPacket( |
| 2333 ch.second->channel(), packet->data(), packet->size()); | 2333 ch.second->channel(), packet->data(), packet->size()); |
| 2334 } | 2334 } |
| 2335 } | 2335 } |
| 2336 | 2336 |
| 2337 void WebRtcVoiceMediaChannel::OnBestConnectionChanged( | |
|
stefan-webrtc
2016/03/17 12:48:19
What will happen when we're not using BUNDLE? It's
honghaiz3
2016/03/23 19:55:40
This is a very good point.
We are using one BWE fo
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| 2338 Connection* best_connection) { | |
| 2339 call_->OnBestConnectionChanged(best_connection); | |
| 2340 } | |
| 2341 | |
| 2337 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { | 2342 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { |
| 2338 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2343 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2339 int channel = GetSendChannelId(ssrc); | 2344 int channel = GetSendChannelId(ssrc); |
| 2340 if (channel == -1) { | 2345 if (channel == -1) { |
| 2341 LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; | 2346 LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; |
| 2342 return false; | 2347 return false; |
| 2343 } | 2348 } |
| 2344 if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) { | 2349 if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) { |
| 2345 LOG_RTCERR2(SetInputMute, channel, muted); | 2350 LOG_RTCERR2(SetInputMute, channel, muted); |
| 2346 return false; | 2351 return false; |
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| 2542 } | 2547 } |
| 2543 } else { | 2548 } else { |
| 2544 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2549 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
| 2545 engine()->voe()->base()->StopPlayout(channel); | 2550 engine()->voe()->base()->StopPlayout(channel); |
| 2546 } | 2551 } |
| 2547 return true; | 2552 return true; |
| 2548 } | 2553 } |
| 2549 } // namespace cricket | 2554 } // namespace cricket |
| 2550 | 2555 |
| 2551 #endif // HAVE_WEBRTC_VOICE | 2556 #endif // HAVE_WEBRTC_VOICE |
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