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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 |
| 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
| 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
| 13 | 13 |
| 14 #include <memory> | 14 #include <memory> |
| 15 #include <string> | 15 #include <string> |
| 16 #include <vector> | 16 #include <vector> |
| 17 | 17 |
| 18 #include "webrtc/api/rtpparameters.h" | 18 #include "webrtc/api/rtpparameters.h" |
| 19 #include "webrtc/api/rtpreceiverinterface.h" | 19 #include "webrtc/api/rtpreceiverinterface.h" |
| 20 #include "webrtc/api/video/video_timing.h" |
| 20 #include "webrtc/base/basictypes.h" | 21 #include "webrtc/base/basictypes.h" |
| 21 #include "webrtc/base/buffer.h" | 22 #include "webrtc/base/buffer.h" |
| 22 #include "webrtc/base/copyonwritebuffer.h" | 23 #include "webrtc/base/copyonwritebuffer.h" |
| 23 #include "webrtc/base/dscp.h" | 24 #include "webrtc/base/dscp.h" |
| 24 #include "webrtc/base/logging.h" | 25 #include "webrtc/base/logging.h" |
| 25 #include "webrtc/base/networkroute.h" | 26 #include "webrtc/base/networkroute.h" |
| 26 #include "webrtc/base/optional.h" | 27 #include "webrtc/base/optional.h" |
| 27 #include "webrtc/base/sigslot.h" | 28 #include "webrtc/base/sigslot.h" |
| 28 #include "webrtc/base/socket.h" | 29 #include "webrtc/base/socket.h" |
| 29 #include "webrtc/base/window.h" | 30 #include "webrtc/base/window.h" |
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| 740 frames_received(0), | 741 frames_received(0), |
| 741 frames_decoded(0), | 742 frames_decoded(0), |
| 742 frames_rendered(0), | 743 frames_rendered(0), |
| 743 decode_ms(0), | 744 decode_ms(0), |
| 744 max_decode_ms(0), | 745 max_decode_ms(0), |
| 745 jitter_buffer_ms(0), | 746 jitter_buffer_ms(0), |
| 746 min_playout_delay_ms(0), | 747 min_playout_delay_ms(0), |
| 747 render_delay_ms(0), | 748 render_delay_ms(0), |
| 748 target_delay_ms(0), | 749 target_delay_ms(0), |
| 749 current_delay_ms(0), | 750 current_delay_ms(0), |
| 750 capture_start_ntp_time_ms(-1) { | 751 capture_start_ntp_time_ms(-1) {} |
| 751 } | |
| 752 | 752 |
| 753 std::vector<SsrcGroup> ssrc_groups; | 753 std::vector<SsrcGroup> ssrc_groups; |
| 754 // TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|? | 754 // TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|? |
| 755 std::string decoder_implementation_name; | 755 std::string decoder_implementation_name; |
| 756 int packets_concealed; | 756 int packets_concealed; |
| 757 int firs_sent; | 757 int firs_sent; |
| 758 int plis_sent; | 758 int plis_sent; |
| 759 int nacks_sent; | 759 int nacks_sent; |
| 760 int frame_width; | 760 int frame_width; |
| 761 int frame_height; | 761 int frame_height; |
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| 786 // Requested latency to account for rendering delay. | 786 // Requested latency to account for rendering delay. |
| 787 int render_delay_ms; | 787 int render_delay_ms; |
| 788 // Target overall delay: network+decode+render, accounting for | 788 // Target overall delay: network+decode+render, accounting for |
| 789 // min_playout_delay_ms. | 789 // min_playout_delay_ms. |
| 790 int target_delay_ms; | 790 int target_delay_ms; |
| 791 // Current overall delay, possibly ramping towards target_delay_ms. | 791 // Current overall delay, possibly ramping towards target_delay_ms. |
| 792 int current_delay_ms; | 792 int current_delay_ms; |
| 793 | 793 |
| 794 // Estimated capture start time in NTP time in ms. | 794 // Estimated capture start time in NTP time in ms. |
| 795 int64_t capture_start_ntp_time_ms; | 795 int64_t capture_start_ntp_time_ms; |
| 796 |
| 797 // Timing frame info: all important timestamps for a full lifetime of a |
| 798 // single 'timing frame'. |
| 799 rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info; |
| 796 }; | 800 }; |
| 797 | 801 |
| 798 struct DataSenderInfo : public MediaSenderInfo { | 802 struct DataSenderInfo : public MediaSenderInfo { |
| 799 DataSenderInfo() | 803 DataSenderInfo() |
| 800 : ssrc(0) { | 804 : ssrc(0) { |
| 801 } | 805 } |
| 802 | 806 |
| 803 uint32_t ssrc; | 807 uint32_t ssrc; |
| 804 }; | 808 }; |
| 805 | 809 |
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| 1206 const char*, | 1210 const char*, |
| 1207 size_t> SignalDataReceived; | 1211 size_t> SignalDataReceived; |
| 1208 // Signal when the media channel is ready to send the stream. Arguments are: | 1212 // Signal when the media channel is ready to send the stream. Arguments are: |
| 1209 // writable(bool) | 1213 // writable(bool) |
| 1210 sigslot::signal1<bool> SignalReadyToSend; | 1214 sigslot::signal1<bool> SignalReadyToSend; |
| 1211 }; | 1215 }; |
| 1212 | 1216 |
| 1213 } // namespace cricket | 1217 } // namespace cricket |
| 1214 | 1218 |
| 1215 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1219 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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