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Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 1827023002: Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added TODO comments regarding SSRC == 0. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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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653 // Estimated capture start time in NTP time in ms. 653 // Estimated capture start time in NTP time in ms.
654 int64_t capture_start_ntp_time_ms; 654 int64_t capture_start_ntp_time_ms;
655 }; 655 };
656 656
657 struct VideoSenderInfo : public MediaSenderInfo { 657 struct VideoSenderInfo : public MediaSenderInfo {
658 VideoSenderInfo() 658 VideoSenderInfo()
659 : packets_cached(0), 659 : packets_cached(0),
660 firs_rcvd(0), 660 firs_rcvd(0),
661 plis_rcvd(0), 661 plis_rcvd(0),
662 nacks_rcvd(0), 662 nacks_rcvd(0),
663 input_frame_width(0),
664 input_frame_height(0),
665 send_frame_width(0), 663 send_frame_width(0),
666 send_frame_height(0), 664 send_frame_height(0),
667 framerate_input(0), 665 framerate_input(0),
668 framerate_sent(0), 666 framerate_sent(0),
669 nominal_bitrate(0), 667 nominal_bitrate(0),
670 preferred_bitrate(0), 668 preferred_bitrate(0),
671 adapt_reason(0), 669 adapt_reason(0),
672 adapt_changes(0), 670 adapt_changes(0),
673 avg_encode_ms(0), 671 avg_encode_ms(0),
674 encode_usage_percent(0) { 672 encode_usage_percent(0) {
675 } 673 }
676 674
677 std::vector<SsrcGroup> ssrc_groups; 675 std::vector<SsrcGroup> ssrc_groups;
678 std::string encoder_implementation_name; 676 std::string encoder_implementation_name;
679 int packets_cached; 677 int packets_cached;
680 int firs_rcvd; 678 int firs_rcvd;
681 int plis_rcvd; 679 int plis_rcvd;
682 int nacks_rcvd; 680 int nacks_rcvd;
683 int input_frame_width;
684 int input_frame_height;
685 int send_frame_width; 681 int send_frame_width;
686 int send_frame_height; 682 int send_frame_height;
687 int framerate_input; 683 int framerate_input;
688 int framerate_sent; 684 int framerate_sent;
689 int nominal_bitrate; 685 int nominal_bitrate;
690 int preferred_bitrate; 686 int preferred_bitrate;
691 int adapt_reason; 687 int adapt_reason;
692 int adapt_changes; 688 int adapt_changes;
693 int avg_encode_ms; 689 int avg_encode_ms;
694 int encode_usage_percent; 690 int encode_usage_percent;
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1122 // Signal when the media channel is ready to send the stream. Arguments are: 1118 // Signal when the media channel is ready to send the stream. Arguments are:
1123 // writable(bool) 1119 // writable(bool)
1124 sigslot::signal1<bool> SignalReadyToSend; 1120 sigslot::signal1<bool> SignalReadyToSend;
1125 // Signal for notifying that the remote side has closed the DataChannel. 1121 // Signal for notifying that the remote side has closed the DataChannel.
1126 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1122 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1127 }; 1123 };
1128 1124
1129 } // namespace cricket 1125 } // namespace cricket
1130 1126
1131 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1127 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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