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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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74 ss << ", sync_group: " << sync_group; | 74 ss << ", sync_group: " << sync_group; |
75 } | 75 } |
76 ss << '}'; | 76 ss << '}'; |
77 return ss.str(); | 77 return ss.str(); |
78 } | 78 } |
79 | 79 |
80 namespace internal { | 80 namespace internal { |
81 AudioReceiveStream::AudioReceiveStream( | 81 AudioReceiveStream::AudioReceiveStream( |
82 CongestionController* congestion_controller, | 82 CongestionController* congestion_controller, |
83 const webrtc::AudioReceiveStream::Config& config, | 83 const webrtc::AudioReceiveStream::Config& config, |
84 const rtc::scoped_refptr<webrtc::AudioState>& audio_state) | 84 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| 85 webrtc::RtcEventLog* event_log) |
85 : config_(config), | 86 : config_(config), |
86 audio_state_(audio_state), | 87 audio_state_(audio_state), |
87 rtp_header_parser_(RtpHeaderParser::Create()) { | 88 rtp_header_parser_(RtpHeaderParser::Create()) { |
88 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); | 89 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
89 RTC_DCHECK_NE(config_.voe_channel_id, -1); | 90 RTC_DCHECK_NE(config_.voe_channel_id, -1); |
90 RTC_DCHECK(audio_state_.get()); | 91 RTC_DCHECK(audio_state_.get()); |
91 RTC_DCHECK(congestion_controller); | 92 RTC_DCHECK(congestion_controller); |
92 RTC_DCHECK(rtp_header_parser_); | 93 RTC_DCHECK(rtp_header_parser_); |
93 | 94 |
94 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); | 95 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
95 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); | 96 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
| 97 channel_proxy_->SetRtcEventLog(event_log); |
96 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); | 98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); |
97 // TODO(solenberg): Config NACK history window (which is a packet count), | 99 // TODO(solenberg): Config NACK history window (which is a packet count), |
98 // using the actual packet size for the configured codec. | 100 // using the actual packet size for the configured codec. |
99 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, | 101 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
100 config_.rtp.nack.rtp_history_ms / 20); | 102 config_.rtp.nack.rtp_history_ms / 20); |
101 | 103 |
102 // TODO(ossu): This is where we'd like to set the decoder factory to | 104 // TODO(ossu): This is where we'd like to set the decoder factory to |
103 // use. However, since it needs to be included when constructing Channel, we | 105 // use. However, since it needs to be included when constructing Channel, we |
104 // cannot do that until we're able to move Channel ownership into the | 106 // cannot do that until we're able to move Channel ownership into the |
105 // Audio{Send,Receive}Streams. The best we can do is check that we're not | 107 // Audio{Send,Receive}Streams. The best we can do is check that we're not |
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137 remote_bitrate_estimator_ = | 139 remote_bitrate_estimator_ = |
138 congestion_controller->GetRemoteBitrateEstimator(true); | 140 congestion_controller->GetRemoteBitrateEstimator(true); |
139 } | 141 } |
140 } | 142 } |
141 | 143 |
142 AudioReceiveStream::~AudioReceiveStream() { | 144 AudioReceiveStream::~AudioReceiveStream() { |
143 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 145 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
144 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); | 146 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
145 channel_proxy_->DeRegisterExternalTransport(); | 147 channel_proxy_->DeRegisterExternalTransport(); |
146 channel_proxy_->ResetCongestionControlObjects(); | 148 channel_proxy_->ResetCongestionControlObjects(); |
| 149 channel_proxy_->SetRtcEventLog(nullptr); |
147 if (remote_bitrate_estimator_) { | 150 if (remote_bitrate_estimator_) { |
148 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); | 151 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); |
149 } | 152 } |
150 } | 153 } |
151 | 154 |
152 void AudioReceiveStream::Start() { | 155 void AudioReceiveStream::Start() { |
153 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 156 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
154 } | 157 } |
155 | 158 |
156 void AudioReceiveStream::Stop() { | 159 void AudioReceiveStream::Stop() { |
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262 | 265 |
263 VoiceEngine* AudioReceiveStream::voice_engine() const { | 266 VoiceEngine* AudioReceiveStream::voice_engine() const { |
264 internal::AudioState* audio_state = | 267 internal::AudioState* audio_state = |
265 static_cast<internal::AudioState*>(audio_state_.get()); | 268 static_cast<internal::AudioState*>(audio_state_.get()); |
266 VoiceEngine* voice_engine = audio_state->voice_engine(); | 269 VoiceEngine* voice_engine = audio_state->voice_engine(); |
267 RTC_DCHECK(voice_engine); | 270 RTC_DCHECK(voice_engine); |
268 return voice_engine; | 271 return voice_engine; |
269 } | 272 } |
270 } // namespace internal | 273 } // namespace internal |
271 } // namespace webrtc | 274 } // namespace webrtc |
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