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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: Address comments from Peter and Taylor. 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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649 // Estimated capture start time in NTP time in ms. 649 // Estimated capture start time in NTP time in ms.
650 int64_t capture_start_ntp_time_ms; 650 int64_t capture_start_ntp_time_ms;
651 }; 651 };
652 652
653 struct VideoSenderInfo : public MediaSenderInfo { 653 struct VideoSenderInfo : public MediaSenderInfo {
654 VideoSenderInfo() 654 VideoSenderInfo()
655 : packets_cached(0), 655 : packets_cached(0),
656 firs_rcvd(0), 656 firs_rcvd(0),
657 plis_rcvd(0), 657 plis_rcvd(0),
658 nacks_rcvd(0), 658 nacks_rcvd(0),
659 input_frame_width(0),
660 input_frame_height(0),
661 send_frame_width(0), 659 send_frame_width(0),
662 send_frame_height(0), 660 send_frame_height(0),
663 framerate_input(0), 661 framerate_input(0),
664 framerate_sent(0), 662 framerate_sent(0),
665 nominal_bitrate(0), 663 nominal_bitrate(0),
666 preferred_bitrate(0), 664 preferred_bitrate(0),
667 adapt_reason(0), 665 adapt_reason(0),
668 adapt_changes(0), 666 adapt_changes(0),
669 avg_encode_ms(0), 667 avg_encode_ms(0),
670 encode_usage_percent(0) { 668 encode_usage_percent(0) {
671 } 669 }
672 670
673 std::vector<SsrcGroup> ssrc_groups; 671 std::vector<SsrcGroup> ssrc_groups;
674 std::string encoder_implementation_name; 672 std::string encoder_implementation_name;
675 int packets_cached; 673 int packets_cached;
676 int firs_rcvd; 674 int firs_rcvd;
677 int plis_rcvd; 675 int plis_rcvd;
678 int nacks_rcvd; 676 int nacks_rcvd;
679 int input_frame_width;
680 int input_frame_height;
681 int send_frame_width; 677 int send_frame_width;
682 int send_frame_height; 678 int send_frame_height;
683 int framerate_input; 679 int framerate_input;
684 int framerate_sent; 680 int framerate_sent;
685 int nominal_bitrate; 681 int nominal_bitrate;
686 int preferred_bitrate; 682 int preferred_bitrate;
687 int adapt_reason; 683 int adapt_reason;
688 int adapt_changes; 684 int adapt_changes;
689 int avg_encode_ms; 685 int avg_encode_ms;
690 int encode_usage_percent; 686 int encode_usage_percent;
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1115 // Signal when the media channel is ready to send the stream. Arguments are: 1111 // Signal when the media channel is ready to send the stream. Arguments are:
1116 // writable(bool) 1112 // writable(bool)
1117 sigslot::signal1<bool> SignalReadyToSend; 1113 sigslot::signal1<bool> SignalReadyToSend;
1118 // Signal for notifying that the remote side has closed the DataChannel. 1114 // Signal for notifying that the remote side has closed the DataChannel.
1119 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1115 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1120 }; 1116 };
1121 1117
1122 } // namespace cricket 1118 } // namespace cricket
1123 1119
1124 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1120 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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