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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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| 764 frame_height(0), | 764 frame_height(0), |
| 765 framerate_rcvd(0), | 765 framerate_rcvd(0), |
| 766 framerate_decoded(0), | 766 framerate_decoded(0), |
| 767 framerate_output(0), | 767 framerate_output(0), |
| 768 framerate_render_input(0), | 768 framerate_render_input(0), |
| 769 framerate_render_output(0), | 769 framerate_render_output(0), |
| 770 frames_received(0), | 770 frames_received(0), |
| 771 frames_decoded(0), | 771 frames_decoded(0), |
| 772 frames_rendered(0), | 772 frames_rendered(0), |
| 773 interframe_delay_max_ms(-1), | 773 interframe_delay_max_ms(-1), |
| 774 content_type(webrtc::VideoContentType::UNSPECIFIED), |
| 774 decode_ms(0), | 775 decode_ms(0), |
| 775 max_decode_ms(0), | 776 max_decode_ms(0), |
| 776 jitter_buffer_ms(0), | 777 jitter_buffer_ms(0), |
| 777 min_playout_delay_ms(0), | 778 min_playout_delay_ms(0), |
| 778 render_delay_ms(0), | 779 render_delay_ms(0), |
| 779 target_delay_ms(0), | 780 target_delay_ms(0), |
| 780 current_delay_ms(0), | 781 current_delay_ms(0), |
| 781 capture_start_ntp_time_ms(-1) {} | 782 capture_start_ntp_time_ms(-1) {} |
| 782 | 783 |
| 783 std::vector<SsrcGroup> ssrc_groups; | 784 std::vector<SsrcGroup> ssrc_groups; |
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| 795 // Framerate as sent to the renderer. | 796 // Framerate as sent to the renderer. |
| 796 int framerate_render_input; | 797 int framerate_render_input; |
| 797 // Framerate that the renderer reports. | 798 // Framerate that the renderer reports. |
| 798 int framerate_render_output; | 799 int framerate_render_output; |
| 799 uint32_t frames_received; | 800 uint32_t frames_received; |
| 800 uint32_t frames_decoded; | 801 uint32_t frames_decoded; |
| 801 uint32_t frames_rendered; | 802 uint32_t frames_rendered; |
| 802 rtc::Optional<uint64_t> qp_sum; | 803 rtc::Optional<uint64_t> qp_sum; |
| 803 int64_t interframe_delay_max_ms; | 804 int64_t interframe_delay_max_ms; |
| 804 | 805 |
| 806 webrtc::VideoContentType content_type; |
| 807 |
| 805 // All stats below are gathered per-VideoReceiver, but some will be correlated | 808 // All stats below are gathered per-VideoReceiver, but some will be correlated |
| 806 // across MediaStreamTracks. NOTE(hta): when sinking stats into per-SSRC | 809 // across MediaStreamTracks. NOTE(hta): when sinking stats into per-SSRC |
| 807 // structures, reflect this in the new layout. | 810 // structures, reflect this in the new layout. |
| 808 | 811 |
| 809 // Current frame decode latency. | 812 // Current frame decode latency. |
| 810 int decode_ms; | 813 int decode_ms; |
| 811 // Maximum observed frame decode latency. | 814 // Maximum observed frame decode latency. |
| 812 int max_decode_ms; | 815 int max_decode_ms; |
| 813 // Jitter (network-related) latency. | 816 // Jitter (network-related) latency. |
| 814 int jitter_buffer_ms; | 817 int jitter_buffer_ms; |
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| 1241 const char*, | 1244 const char*, |
| 1242 size_t> SignalDataReceived; | 1245 size_t> SignalDataReceived; |
| 1243 // Signal when the media channel is ready to send the stream. Arguments are: | 1246 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1244 // writable(bool) | 1247 // writable(bool) |
| 1245 sigslot::signal1<bool> SignalReadyToSend; | 1248 sigslot::signal1<bool> SignalReadyToSend; |
| 1246 }; | 1249 }; |
| 1247 | 1250 |
| 1248 } // namespace cricket | 1251 } // namespace cricket |
| 1249 | 1252 |
| 1250 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1253 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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