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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_receive_stream.h" | 11 #include "webrtc/audio/audio_receive_stream.h" |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 #include <utility> | 14 #include <utility> |
15 | 15 |
16 #include "webrtc/api/call/audio_sink.h" | 16 #include "webrtc/api/call/audio_sink.h" |
17 #include "webrtc/audio/audio_send_stream.h" | 17 #include "webrtc/audio/audio_send_stream.h" |
18 #include "webrtc/audio/audio_state.h" | 18 #include "webrtc/audio/audio_state.h" |
19 #include "webrtc/audio/conversion.h" | 19 #include "webrtc/audio/conversion.h" |
20 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
21 #include "webrtc/base/logging.h" | 21 #include "webrtc/base/logging.h" |
22 #include "webrtc/base/timeutils.h" | 22 #include "webrtc/base/timeutils.h" |
23 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | |
24 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" | 23 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" |
25 #include "webrtc/voice_engine/channel_proxy.h" | 24 #include "webrtc/voice_engine/channel_proxy.h" |
26 #include "webrtc/voice_engine/include/voe_base.h" | 25 #include "webrtc/voice_engine/include/voe_base.h" |
27 #include "webrtc/voice_engine/include/voe_codec.h" | 26 #include "webrtc/voice_engine/include/voe_codec.h" |
28 #include "webrtc/voice_engine/include/voe_neteq_stats.h" | 27 #include "webrtc/voice_engine/include/voe_neteq_stats.h" |
29 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
30 #include "webrtc/voice_engine/include/voe_video_sync.h" | 29 #include "webrtc/voice_engine/include/voe_video_sync.h" |
31 #include "webrtc/voice_engine/include/voe_volume_control.h" | 30 #include "webrtc/voice_engine/include/voe_volume_control.h" |
32 #include "webrtc/voice_engine/voice_engine_impl.h" | 31 #include "webrtc/voice_engine/voice_engine_impl.h" |
33 | 32 |
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73 ss << ", voe_channel_id: " << voe_channel_id; | 72 ss << ", voe_channel_id: " << voe_channel_id; |
74 if (!sync_group.empty()) { | 73 if (!sync_group.empty()) { |
75 ss << ", sync_group: " << sync_group; | 74 ss << ", sync_group: " << sync_group; |
76 } | 75 } |
77 ss << '}'; | 76 ss << '}'; |
78 return ss.str(); | 77 return ss.str(); |
79 } | 78 } |
80 | 79 |
81 namespace internal { | 80 namespace internal { |
82 AudioReceiveStream::AudioReceiveStream( | 81 AudioReceiveStream::AudioReceiveStream( |
83 CongestionController* congestion_controller, | 82 PacketRouter* packet_router, |
83 RemoteBitrateEstimator* remote_bitrate_estimator, | |
84 const webrtc::AudioReceiveStream::Config& config, | 84 const webrtc::AudioReceiveStream::Config& config, |
85 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, | 85 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
86 webrtc::RtcEventLog* event_log) | 86 webrtc::RtcEventLog* event_log) |
87 : config_(config), | 87 : config_(config), |
88 audio_state_(audio_state), | 88 audio_state_(audio_state), |
89 rtp_header_parser_(RtpHeaderParser::Create()) { | 89 rtp_header_parser_(RtpHeaderParser::Create()) { |
90 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); | 90 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
91 RTC_DCHECK_NE(config_.voe_channel_id, -1); | 91 RTC_DCHECK_NE(config_.voe_channel_id, -1); |
92 RTC_DCHECK(audio_state_.get()); | 92 RTC_DCHECK(audio_state_.get()); |
93 RTC_DCHECK(congestion_controller); | 93 RTC_DCHECK(packet_router); |
the sun
2016/11/21 13:55:31
Can we DCHECK(remote_bitrate_estimator) too?
nisse-webrtc
2016/11/21 14:07:36
Not sure if that's appropriate, it's used only if
the sun
2016/11/21 15:36:11
If we can avoid having to handle the null case, th
nisse-webrtc
2016/11/23 08:11:44
In this cl, the call site looks like this:
Audi
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94 RTC_DCHECK(rtp_header_parser_); | 94 RTC_DCHECK(rtp_header_parser_); |
95 | 95 |
96 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); | 96 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
97 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); | 97 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
98 channel_proxy_->SetRtcEventLog(event_log); | 98 channel_proxy_->SetRtcEventLog(event_log); |
99 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); | 99 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); |
100 // TODO(solenberg): Config NACK history window (which is a packet count), | 100 // TODO(solenberg): Config NACK history window (which is a packet count), |
101 // using the actual packet size for the configured codec. | 101 // using the actual packet size for the configured codec. |
102 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, | 102 channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
103 config_.rtp.nack.rtp_history_ms / 20); | 103 config_.rtp.nack.rtp_history_ms / 20); |
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122 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { | 122 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
123 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); | 123 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); |
124 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( | 124 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
125 kRtpExtensionTransportSequenceNumber, extension.id); | 125 kRtpExtensionTransportSequenceNumber, extension.id); |
126 RTC_DCHECK(registered); | 126 RTC_DCHECK(registered); |
127 } else { | 127 } else { |
128 RTC_NOTREACHED() << "Unsupported RTP extension."; | 128 RTC_NOTREACHED() << "Unsupported RTP extension."; |
129 } | 129 } |
130 } | 130 } |
131 // Configure bandwidth estimation. | 131 // Configure bandwidth estimation. |
132 channel_proxy_->RegisterReceiverCongestionControlObjects( | 132 channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router); |
133 congestion_controller->packet_router()); | |
134 if (UseSendSideBwe(config)) { | 133 if (UseSendSideBwe(config)) { |
135 remote_bitrate_estimator_ = | 134 remote_bitrate_estimator_ = remote_bitrate_estimator; |
nisse-webrtc
2016/11/21 14:07:36
A DCHECK(remote_bitrate_estimator) might make more
the sun
2016/11/21 15:36:11
If Call might be giving us a null in this situatio
nisse-webrtc
2016/11/23 08:11:44
Currently we never get null. Motivation for a DCHE
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136 congestion_controller->GetRemoteBitrateEstimator(true); | |
137 } | 135 } |
138 } | 136 } |
139 | 137 |
140 AudioReceiveStream::~AudioReceiveStream() { | 138 AudioReceiveStream::~AudioReceiveStream() { |
141 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 139 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
142 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); | 140 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
143 Stop(); | 141 Stop(); |
144 channel_proxy_->DisassociateSendChannel(); | 142 channel_proxy_->DisassociateSendChannel(); |
145 channel_proxy_->DeRegisterExternalTransport(); | 143 channel_proxy_->DeRegisterExternalTransport(); |
146 channel_proxy_->ResetCongestionControlObjects(); | 144 channel_proxy_->ResetCongestionControlObjects(); |
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298 | 296 |
299 VoiceEngine* AudioReceiveStream::voice_engine() const { | 297 VoiceEngine* AudioReceiveStream::voice_engine() const { |
300 internal::AudioState* audio_state = | 298 internal::AudioState* audio_state = |
301 static_cast<internal::AudioState*>(audio_state_.get()); | 299 static_cast<internal::AudioState*>(audio_state_.get()); |
302 VoiceEngine* voice_engine = audio_state->voice_engine(); | 300 VoiceEngine* voice_engine = audio_state->voice_engine(); |
303 RTC_DCHECK(voice_engine); | 301 RTC_DCHECK(voice_engine); |
304 return voice_engine; | 302 return voice_engine; |
305 } | 303 } |
306 } // namespace internal | 304 } // namespace internal |
307 } // namespace webrtc | 305 } // namespace webrtc |
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