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Issue 2663063008: Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ (Closed)
Patch Set: Created 3 years, 10 months ago
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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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