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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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41 } // namespace | 41 } // namespace |
42 | 42 |
43 namespace internal { | 43 namespace internal { |
44 AudioSendStream::AudioSendStream( | 44 AudioSendStream::AudioSendStream( |
45 const webrtc::AudioSendStream::Config& config, | 45 const webrtc::AudioSendStream::Config& config, |
46 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 46 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
47 rtc::TaskQueue* worker_queue, | 47 rtc::TaskQueue* worker_queue, |
48 PacketRouter* packet_router, | 48 PacketRouter* packet_router, |
49 CongestionController* congestion_controller, | 49 CongestionController* congestion_controller, |
50 BitrateAllocator* bitrate_allocator, | 50 BitrateAllocator* bitrate_allocator, |
51 RtcEventLog* event_log) | 51 RtcEventLog* event_log, |
| 52 RtcpRttStats* rtcp_rtt_stats) |
52 : worker_queue_(worker_queue), | 53 : worker_queue_(worker_queue), |
53 config_(config), | 54 config_(config), |
54 audio_state_(audio_state), | 55 audio_state_(audio_state), |
55 bitrate_allocator_(bitrate_allocator) { | 56 bitrate_allocator_(bitrate_allocator) { |
56 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); | 57 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); |
57 RTC_DCHECK_NE(config_.voe_channel_id, -1); | 58 RTC_DCHECK_NE(config_.voe_channel_id, -1); |
58 RTC_DCHECK(audio_state_.get()); | 59 RTC_DCHECK(audio_state_.get()); |
59 RTC_DCHECK(congestion_controller); | 60 RTC_DCHECK(congestion_controller); |
60 | 61 |
61 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); | 62 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
62 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); | 63 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
63 channel_proxy_->SetRtcEventLog(event_log); | 64 channel_proxy_->SetRtcEventLog(event_log); |
| 65 channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats); |
64 channel_proxy_->RegisterSenderCongestionControlObjects( | 66 channel_proxy_->RegisterSenderCongestionControlObjects( |
65 congestion_controller->pacer(), | 67 congestion_controller->pacer(), |
66 congestion_controller->GetTransportFeedbackObserver(), packet_router); | 68 congestion_controller->GetTransportFeedbackObserver(), packet_router); |
67 channel_proxy_->SetRTCPStatus(true); | 69 channel_proxy_->SetRTCPStatus(true); |
68 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); | 70 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); |
69 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); | 71 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); |
70 // TODO(solenberg): Config NACK history window (which is a packet count), | 72 // TODO(solenberg): Config NACK history window (which is a packet count), |
71 // using the actual packet size for the configured codec. | 73 // using the actual packet size for the configured codec. |
72 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, | 74 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
73 config_.rtp.nack.rtp_history_ms / 20); | 75 config_.rtp.nack.rtp_history_ms / 20); |
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87 LOG(LS_ERROR) << "Failed to set up send codec state."; | 89 LOG(LS_ERROR) << "Failed to set up send codec state."; |
88 } | 90 } |
89 } | 91 } |
90 | 92 |
91 AudioSendStream::~AudioSendStream() { | 93 AudioSendStream::~AudioSendStream() { |
92 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 94 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
93 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); | 95 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); |
94 channel_proxy_->DeRegisterExternalTransport(); | 96 channel_proxy_->DeRegisterExternalTransport(); |
95 channel_proxy_->ResetCongestionControlObjects(); | 97 channel_proxy_->ResetCongestionControlObjects(); |
96 channel_proxy_->SetRtcEventLog(nullptr); | 98 channel_proxy_->SetRtcEventLog(nullptr); |
| 99 channel_proxy_->SetRtcpRttStats(nullptr); |
97 } | 100 } |
98 | 101 |
99 void AudioSendStream::Start() { | 102 void AudioSendStream::Start() { |
100 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 103 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
101 if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) { | 104 if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) { |
102 RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps); | 105 RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps); |
103 rtc::Event thread_sync_event(false /* manual_reset */, false); | 106 rtc::Event thread_sync_event(false /* manual_reset */, false); |
104 worker_queue_->PostTask([this, &thread_sync_event] { | 107 worker_queue_->PostTask([this, &thread_sync_event] { |
105 bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps, | 108 bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps, |
106 config_.max_bitrate_bps, 0, true); | 109 config_.max_bitrate_bps, 0, true); |
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378 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); | 381 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); |
379 return false; | 382 return false; |
380 } | 383 } |
381 } | 384 } |
382 } | 385 } |
383 return true; | 386 return true; |
384 } | 387 } |
385 | 388 |
386 } // namespace internal | 389 } // namespace internal |
387 } // namespace webrtc | 390 } // namespace webrtc |
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