Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(82)

Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.h

Issue 1814233002: Delete default_send_ssrc_. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Delete ssrc==0 check in RemoveSendStream. Created 4 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
« no previous file with comments | « webrtc/media/base/videoengine_unittest.h ('k') | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
(...skipping 154 matching lines...) Expand 10 before | Expand all | Expand 10 after
165 165
166 void OnPacketReceived(rtc::Buffer* packet, 166 void OnPacketReceived(rtc::Buffer* packet,
167 const rtc::PacketTime& packet_time) override; 167 const rtc::PacketTime& packet_time) override;
168 void OnRtcpReceived(rtc::Buffer* packet, 168 void OnRtcpReceived(rtc::Buffer* packet,
169 const rtc::PacketTime& packet_time) override; 169 const rtc::PacketTime& packet_time) override;
170 void OnReadyToSend(bool ready) override; 170 void OnReadyToSend(bool ready) override;
171 void SetInterface(NetworkInterface* iface) override; 171 void SetInterface(NetworkInterface* iface) override;
172 172
173 // Implemented for VideoMediaChannelTest. 173 // Implemented for VideoMediaChannelTest.
174 bool sending() const { return sending_; } 174 bool sending() const { return sending_; }
175 uint32_t GetDefaultSendChannelSsrc() { return default_send_ssrc_; }
176 175
177 private: 176 private:
178 class WebRtcVideoReceiveStream; 177 class WebRtcVideoReceiveStream;
179 struct VideoCodecSettings { 178 struct VideoCodecSettings {
180 VideoCodecSettings(); 179 VideoCodecSettings();
181 180
182 bool operator==(const VideoCodecSettings& other) const; 181 bool operator==(const VideoCodecSettings& other) const;
183 bool operator!=(const VideoCodecSettings& other) const; 182 bool operator!=(const VideoCodecSettings& other) const;
184 183
185 VideoCodec codec; 184 VideoCodec codec;
(...skipping 311 matching lines...) Expand 10 before | Expand all | Expand 10 after
497 void FillReceiverStats(VideoMediaInfo* info); 496 void FillReceiverStats(VideoMediaInfo* info);
498 void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats, 497 void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
499 VideoMediaInfo* info); 498 VideoMediaInfo* info);
500 499
501 rtc::ThreadChecker thread_checker_; 500 rtc::ThreadChecker thread_checker_;
502 501
503 uint32_t rtcp_receiver_report_ssrc_; 502 uint32_t rtcp_receiver_report_ssrc_;
504 bool sending_; 503 bool sending_;
505 webrtc::Call* const call_; 504 webrtc::Call* const call_;
506 505
507 uint32_t default_send_ssrc_;
508
509 DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_; 506 DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
510 UnsignalledSsrcHandler* const unsignalled_ssrc_handler_; 507 UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
511 508
512 const MediaConfig::Video video_config_; 509 const MediaConfig::Video video_config_;
513 510
514 rtc::CriticalSection stream_crit_; 511 rtc::CriticalSection stream_crit_;
515 // Using primary-ssrc (first ssrc) as key. 512 // Using primary-ssrc (first ssrc) as key.
516 std::map<uint32_t, WebRtcVideoSendStream*> send_streams_ 513 std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
517 GUARDED_BY(stream_crit_); 514 GUARDED_BY(stream_crit_);
518 std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_ 515 std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
(...skipping 12 matching lines...) Expand all
531 // TODO(deadbeef): Don't duplicate information between 528 // TODO(deadbeef): Don't duplicate information between
532 // send_params/recv_params, rtp_extensions, options, etc. 529 // send_params/recv_params, rtp_extensions, options, etc.
533 VideoSendParameters send_params_; 530 VideoSendParameters send_params_;
534 VideoOptions default_send_options_; 531 VideoOptions default_send_options_;
535 VideoRecvParameters recv_params_; 532 VideoRecvParameters recv_params_;
536 }; 533 };
537 534
538 } // namespace cricket 535 } // namespace cricket
539 536
540 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 537 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
OLDNEW
« no previous file with comments | « webrtc/media/base/videoengine_unittest.h ('k') | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698