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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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| 61 if (!sync_group.empty()) { | 61 if (!sync_group.empty()) { |
| 62 ss << ", sync_group: " << sync_group; | 62 ss << ", sync_group: " << sync_group; |
| 63 } | 63 } |
| 64 ss << '}'; | 64 ss << '}'; |
| 65 return ss.str(); | 65 return ss.str(); |
| 66 } | 66 } |
| 67 | 67 |
| 68 namespace internal { | 68 namespace internal { |
| 69 AudioReceiveStream::AudioReceiveStream( | 69 AudioReceiveStream::AudioReceiveStream( |
| 70 PacketRouter* packet_router, | 70 PacketRouter* packet_router, |
| 71 RemoteBitrateEstimator* remote_bitrate_estimator, | |
| 72 const webrtc::AudioReceiveStream::Config& config, | 71 const webrtc::AudioReceiveStream::Config& config, |
| 73 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 72 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| 74 webrtc::RtcEventLog* event_log) | 73 webrtc::RtcEventLog* event_log) |
| 75 : remote_bitrate_estimator_(remote_bitrate_estimator), | 74 : config_(config), |
| 76 config_(config), | |
| 77 audio_state_(audio_state), | 75 audio_state_(audio_state), |
| 78 rtp_header_parser_(RtpHeaderParser::Create()) { | 76 rtp_header_parser_(RtpHeaderParser::Create()) { |
| 79 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); | 77 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
| 80 RTC_DCHECK_NE(config_.voe_channel_id, -1); | 78 RTC_DCHECK_NE(config_.voe_channel_id, -1); |
| 81 RTC_DCHECK(audio_state_.get()); | 79 RTC_DCHECK(audio_state_.get()); |
| 82 RTC_DCHECK(packet_router); | 80 RTC_DCHECK(packet_router); |
| 83 RTC_DCHECK(remote_bitrate_estimator); | |
| 84 RTC_DCHECK(rtp_header_parser_); | 81 RTC_DCHECK(rtp_header_parser_); |
| 85 | 82 |
| 86 module_process_thread_checker_.DetachFromThread(); | 83 module_process_thread_checker_.DetachFromThread(); |
| 87 | 84 |
| 88 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); | 85 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
| 89 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); | 86 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
| 90 channel_proxy_->SetRtcEventLog(event_log); | 87 channel_proxy_->SetRtcEventLog(event_log); |
| 91 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); | 88 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); |
| 92 // TODO(solenberg): Config NACK history window (which is a packet count), | 89 // TODO(solenberg): Config NACK history window (which is a packet count), |
| 93 // using the actual packet size for the configured codec. | 90 // using the actual packet size for the configured codec. |
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| 131 AudioReceiveStream::~AudioReceiveStream() { | 128 AudioReceiveStream::~AudioReceiveStream() { |
| 132 RTC_DCHECK_RUN_ON(&worker_thread_checker_); | 129 RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| 133 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); | 130 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
| 134 if (playing_) { | 131 if (playing_) { |
| 135 Stop(); | 132 Stop(); |
| 136 } | 133 } |
| 137 channel_proxy_->DisassociateSendChannel(); | 134 channel_proxy_->DisassociateSendChannel(); |
| 138 channel_proxy_->DeRegisterExternalTransport(); | 135 channel_proxy_->DeRegisterExternalTransport(); |
| 139 channel_proxy_->ResetCongestionControlObjects(); | 136 channel_proxy_->ResetCongestionControlObjects(); |
| 140 channel_proxy_->SetRtcEventLog(nullptr); | 137 channel_proxy_->SetRtcEventLog(nullptr); |
| 141 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); | |
| 142 } | 138 } |
| 143 | 139 |
| 144 void AudioReceiveStream::Start() { | 140 void AudioReceiveStream::Start() { |
| 145 RTC_DCHECK_RUN_ON(&worker_thread_checker_); | 141 RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
| 146 if (playing_) { | 142 if (playing_) { |
| 147 return; | 143 return; |
| 148 } | 144 } |
| 149 | 145 |
| 150 int error = SetVoiceEnginePlayout(true); | 146 int error = SetVoiceEnginePlayout(true); |
| 151 if (error != 0) { | 147 if (error != 0) { |
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| 353 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { | 349 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { |
| 354 ScopedVoEInterface<VoEBase> base(voice_engine()); | 350 ScopedVoEInterface<VoEBase> base(voice_engine()); |
| 355 if (playout) { | 351 if (playout) { |
| 356 return base->StartPlayout(config_.voe_channel_id); | 352 return base->StartPlayout(config_.voe_channel_id); |
| 357 } else { | 353 } else { |
| 358 return base->StopPlayout(config_.voe_channel_id); | 354 return base->StopPlayout(config_.voe_channel_id); |
| 359 } | 355 } |
| 360 } | 356 } |
| 361 } // namespace internal | 357 } // namespace internal |
| 362 } // namespace webrtc | 358 } // namespace webrtc |
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