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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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406 } | 406 } |
407 | 407 |
408 int32_t ModuleRtpRtcpImpl::SendOutgoingData( | 408 int32_t ModuleRtpRtcpImpl::SendOutgoingData( |
409 FrameType frame_type, | 409 FrameType frame_type, |
410 int8_t payload_type, | 410 int8_t payload_type, |
411 uint32_t time_stamp, | 411 uint32_t time_stamp, |
412 int64_t capture_time_ms, | 412 int64_t capture_time_ms, |
413 const uint8_t* payload_data, | 413 const uint8_t* payload_data, |
414 size_t payload_size, | 414 size_t payload_size, |
415 const RTPFragmentationHeader* fragmentation, | 415 const RTPFragmentationHeader* fragmentation, |
416 const RTPVideoHeader* rtp_video_hdr) { | 416 const RTPVideoHeader* rtp_video_hdr, |
| 417 uint32_t* frame_id_out) { |
417 rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms); | 418 rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms); |
418 // Make sure an RTCP report isn't queued behind a key frame. | 419 // Make sure an RTCP report isn't queued behind a key frame. |
419 if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) { | 420 if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) { |
420 rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); | 421 rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); |
421 } | 422 } |
422 return rtp_sender_.SendOutgoingData( | 423 return rtp_sender_.SendOutgoingData( |
423 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, | 424 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, |
424 payload_size, fragmentation, rtp_video_hdr); | 425 payload_size, fragmentation, rtp_video_hdr, frame_id_out); |
425 } | 426 } |
426 | 427 |
427 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, | 428 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, |
428 uint16_t sequence_number, | 429 uint16_t sequence_number, |
429 int64_t capture_time_ms, | 430 int64_t capture_time_ms, |
430 bool retransmission, | 431 bool retransmission, |
431 int probe_cluster_id) { | 432 int probe_cluster_id) { |
432 if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { | 433 if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { |
433 return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, | 434 return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, |
434 retransmission, probe_cluster_id); | 435 retransmission, probe_cluster_id); |
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989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( | 990 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( |
990 StreamDataCountersCallback* callback) { | 991 StreamDataCountersCallback* callback) { |
991 rtp_sender_.RegisterRtpStatisticsCallback(callback); | 992 rtp_sender_.RegisterRtpStatisticsCallback(callback); |
992 } | 993 } |
993 | 994 |
994 StreamDataCountersCallback* | 995 StreamDataCountersCallback* |
995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 996 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
996 return rtp_sender_.GetRtpStatisticsCallback(); | 997 return rtp_sender_.GetRtpStatisticsCallback(); |
997 } | 998 } |
998 } // namespace webrtc | 999 } // namespace webrtc |
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