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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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| 200 ReceiveRtpConfig(const std::vector<RtpExtension>& extensions, | 200 ReceiveRtpConfig(const std::vector<RtpExtension>& extensions, |
| 201 bool transport_cc) | 201 bool transport_cc) |
| 202 : extensions(extensions), transport_cc(transport_cc) {} | 202 : extensions(extensions), transport_cc(transport_cc) {} |
| 203 | 203 |
| 204 // Registered RTP header extensions for each stream. Note that RTP header | 204 // Registered RTP header extensions for each stream. Note that RTP header |
| 205 // extensions are negotiated per track ("m= line") in the SDP, but we have | 205 // extensions are negotiated per track ("m= line") in the SDP, but we have |
| 206 // no notion of tracks at the Call level. We therefore store the RTP header | 206 // no notion of tracks at the Call level. We therefore store the RTP header |
| 207 // extensions per SSRC instead, which leads to some storage overhead. | 207 // extensions per SSRC instead, which leads to some storage overhead. |
| 208 RtpHeaderExtensionMap extensions; | 208 RtpHeaderExtensionMap extensions; |
| 209 // Set if the RTCP feedback message needed for send side BWE was negotiated. | 209 // Set if the RTCP feedback message needed for send side BWE was negotiated. |
| 210 bool transport_cc; | 210 bool transport_cc = false; |
|
nisse-webrtc
2017/02/01 14:50:17
Does style guide suggest always providing default
the sun
2017/02/01 15:12:06
I don't know. It's a habit that has saved me many
nisse-webrtc
2017/02/02 08:10:11
Ok.
| |
| 211 }; | 211 }; |
| 212 std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_ | 212 std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_ |
| 213 GUARDED_BY(receive_crit_); | 213 GUARDED_BY(receive_crit_); |
| 214 | 214 |
| 215 std::unique_ptr<RWLockWrapper> send_crit_; | 215 std::unique_ptr<RWLockWrapper> send_crit_; |
| 216 // Audio and Video send streams are owned by the client that creates them. | 216 // Audio and Video send streams are owned by the client that creates them. |
| 217 std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_); | 217 std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_); |
| 218 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_); | 218 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_); |
| 219 std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_); | 219 std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_); |
| 220 | 220 |
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| 1245 if (transport_cc != header.extension.hasTransportSequenceNumber) { | 1245 if (transport_cc != header.extension.hasTransportSequenceNumber) { |
| 1246 LOG(LS_ERROR) << "Inconsistent configuration of send side BWE."; | 1246 LOG(LS_ERROR) << "Inconsistent configuration of send side BWE."; |
| 1247 return; | 1247 return; |
| 1248 } | 1248 } |
| 1249 congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(), | 1249 congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(), |
| 1250 packet.payload_size(), header); | 1250 packet.payload_size(), header); |
| 1251 } | 1251 } |
| 1252 | 1252 |
| 1253 } // namespace internal | 1253 } // namespace internal |
| 1254 } // namespace webrtc | 1254 } // namespace webrtc |
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