Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(260)

Side by Side Diff: webrtc/video/rtp_stream_receiver.cc

Issue 2729053002: Add location to RegisterModule (Closed)
Patch Set: Format BUILD.gn Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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/video/rtp_stream_receiver.h" 11 #include "webrtc/video/rtp_stream_receiver.h"
12 12
13 #include <vector> 13 #include <vector>
14 #include <utility> 14 #include <utility>
15 15
16 #include "webrtc/base/checks.h" 16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/location.h"
17 #include "webrtc/base/logging.h" 18 #include "webrtc/base/logging.h"
18 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
19 #include "webrtc/config.h" 20 #include "webrtc/config.h"
20 #include "webrtc/media/base/mediaconstants.h" 21 #include "webrtc/media/base/mediaconstants.h"
21 #include "webrtc/modules/pacing/packet_router.h" 22 #include "webrtc/modules/pacing/packet_router.h"
22 #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"
23 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 24 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
24 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" 25 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
25 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 26 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
26 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 27 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after
177 config_.rtp.ulpfec.red_payload_type); 178 config_.rtp.ulpfec.red_payload_type);
178 } 179 }
179 } 180 }
180 181
181 if (config_.rtp.rtcp_xr.receiver_reference_time_report) 182 if (config_.rtp.rtcp_xr.receiver_reference_time_report)
182 rtp_rtcp_->SetRtcpXrRrtrStatus(true); 183 rtp_rtcp_->SetRtcpXrRrtrStatus(true);
183 184
184 // Stats callback for CNAME changes. 185 // Stats callback for CNAME changes.
185 rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy); 186 rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
186 187
187 process_thread_->RegisterModule(rtp_rtcp_.get()); 188 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
188 189
189 if (config_.rtp.nack.rtp_history_ms != 0) { 190 if (config_.rtp.nack.rtp_history_ms != 0) {
190 nack_module_.reset( 191 nack_module_.reset(
191 new NackModule(clock_, nack_sender, keyframe_request_sender)); 192 new NackModule(clock_, nack_sender, keyframe_request_sender));
192 process_thread_->RegisterModule(nack_module_.get()); 193 process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
193 } 194 }
194 195
195 packet_buffer_ = video_coding::PacketBuffer::Create( 196 packet_buffer_ = video_coding::PacketBuffer::Create(
196 clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this); 197 clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this);
197 reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this)); 198 reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this));
198 } 199 }
199 200
200 RtpStreamReceiver::~RtpStreamReceiver() { 201 RtpStreamReceiver::~RtpStreamReceiver() {
201 if (nack_module_) { 202 if (nack_module_) {
202 process_thread_->DeRegisterModule(nack_module_.get()); 203 process_thread_->DeRegisterModule(nack_module_.get());
(...skipping 453 matching lines...) Expand 10 before | Expand all | Expand 10 after
656 return; 657 return;
657 658
658 if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str())) 659 if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
659 return; 660 return;
660 661
661 tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(), 662 tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
662 sprop_decoder.pps_nalu()); 663 sprop_decoder.pps_nalu());
663 } 664 }
664 665
665 } // namespace webrtc 666 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/video_coding/jitter_buffer_unittest.cc ('k') | webrtc/video/video_receive_stream.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698