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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 |
11 #include "webrtc/audio/audio_send_stream.h" | 11 #include "webrtc/audio/audio_send_stream.h" |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "webrtc/audio/audio_state.h" | 15 #include "webrtc/audio/audio_state.h" |
16 #include "webrtc/audio/conversion.h" | 16 #include "webrtc/audio/conversion.h" |
17 #include "webrtc/audio/scoped_voe_interface.h" | 17 #include "webrtc/audio/scoped_voe_interface.h" |
18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
19 #include "webrtc/base/event.h" | 19 #include "webrtc/base/event.h" |
20 #include "webrtc/base/logging.h" | 20 #include "webrtc/base/logging.h" |
21 #include "webrtc/base/task_queue.h" | 21 #include "webrtc/base/task_queue.h" |
| 22 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
22 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | 23 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
23 #include "webrtc/modules/pacing/paced_sender.h" | 24 #include "webrtc/modules/pacing/paced_sender.h" |
24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
25 #include "webrtc/voice_engine/channel_proxy.h" | 26 #include "webrtc/voice_engine/channel_proxy.h" |
26 #include "webrtc/voice_engine/include/voe_base.h" | 27 #include "webrtc/voice_engine/include/voe_base.h" |
27 #include "webrtc/voice_engine/include/voe_volume_control.h" | 28 #include "webrtc/voice_engine/include/voe_volume_control.h" |
28 #include "webrtc/voice_engine/voice_engine_impl.h" | 29 #include "webrtc/voice_engine/voice_engine_impl.h" |
29 | 30 |
30 namespace webrtc { | 31 namespace webrtc { |
31 | 32 |
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55 congestion_controller_(congestion_controller) { | 56 congestion_controller_(congestion_controller) { |
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); |
64 channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats); | 65 channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats); |
65 channel_proxy_->RegisterSenderCongestionControlObjects( | |
66 congestion_controller->pacer(), | |
67 congestion_controller->GetTransportFeedbackObserver(), packet_router); | |
68 channel_proxy_->SetRTCPStatus(true); | 66 channel_proxy_->SetRTCPStatus(true); |
69 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); | 67 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); |
70 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); | 68 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); |
71 // TODO(solenberg): Config NACK history window (which is a packet count), | 69 // TODO(solenberg): Config NACK history window (which is a packet count), |
72 // using the actual packet size for the configured codec. | 70 // using the actual packet size for the configured codec. |
73 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, | 71 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
74 config_.rtp.nack.rtp_history_ms / 20); | 72 config_.rtp.nack.rtp_history_ms / 20); |
75 | 73 |
76 channel_proxy_->RegisterExternalTransport(config.send_transport); | 74 channel_proxy_->RegisterExternalTransport(config.send_transport); |
77 | 75 |
78 for (const auto& extension : config.rtp.extensions) { | 76 for (const auto& extension : config.rtp.extensions) { |
79 if (extension.uri == RtpExtension::kAudioLevelUri) { | 77 if (extension.uri == RtpExtension::kAudioLevelUri) { |
80 channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id); | 78 channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id); |
81 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { | 79 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
82 channel_proxy_->EnableSendTransportSequenceNumber(extension.id); | 80 channel_proxy_->EnableSendTransportSequenceNumber(extension.id); |
83 congestion_controller->EnablePeriodicAlrProbing(true); | 81 congestion_controller->EnablePeriodicAlrProbing(true); |
| 82 bandwidth_observer_.reset(congestion_controller->GetBitrateController() |
| 83 ->CreateRtcpBandwidthObserver()); |
84 } else { | 84 } else { |
85 RTC_NOTREACHED() << "Registering unsupported RTP extension."; | 85 RTC_NOTREACHED() << "Registering unsupported RTP extension."; |
86 } | 86 } |
87 } | 87 } |
| 88 channel_proxy_->RegisterSenderCongestionControlObjects( |
| 89 congestion_controller->pacer(), |
| 90 congestion_controller->GetTransportFeedbackObserver(), packet_router, |
| 91 bandwidth_observer_.get()); |
88 if (!SetupSendCodec()) { | 92 if (!SetupSendCodec()) { |
89 LOG(LS_ERROR) << "Failed to set up send codec state."; | 93 LOG(LS_ERROR) << "Failed to set up send codec state."; |
90 } | 94 } |
91 } | 95 } |
92 | 96 |
93 AudioSendStream::~AudioSendStream() { | 97 AudioSendStream::~AudioSendStream() { |
94 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 98 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
95 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); | 99 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); |
96 channel_proxy_->DeRegisterExternalTransport(); | 100 channel_proxy_->DeRegisterExternalTransport(); |
97 channel_proxy_->ResetCongestionControlObjects(); | 101 channel_proxy_->ResetCongestionControlObjects(); |
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375 LOG(LS_WARNING) << "SetVADStatus() failed."; | 379 LOG(LS_WARNING) << "SetVADStatus() failed."; |
376 return false; | 380 return false; |
377 } | 381 } |
378 } | 382 } |
379 } | 383 } |
380 return true; | 384 return true; |
381 } | 385 } |
382 | 386 |
383 } // namespace internal | 387 } // namespace internal |
384 } // namespace webrtc | 388 } // namespace webrtc |
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