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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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387 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, | 387 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, |
388 payload_size, fragmentation, rtp_video_header, transport_frame_id_out); | 388 payload_size, fragmentation, rtp_video_header, transport_frame_id_out); |
389 } | 389 } |
390 | 390 |
391 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, | 391 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, |
392 uint16_t sequence_number, | 392 uint16_t sequence_number, |
393 int64_t capture_time_ms, | 393 int64_t capture_time_ms, |
394 bool retransmission, | 394 bool retransmission, |
395 const PacedPacketInfo& pacing_info) { | 395 const PacedPacketInfo& pacing_info) { |
396 return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms, | 396 return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms, |
397 retransmission, | 397 retransmission, pacing_info); |
398 pacing_info.probe_cluster_id); | |
399 } | 398 } |
400 | 399 |
401 size_t ModuleRtpRtcpImpl::TimeToSendPadding( | 400 size_t ModuleRtpRtcpImpl::TimeToSendPadding( |
402 size_t bytes, | 401 size_t bytes, |
403 const PacedPacketInfo& pacing_info) { | 402 const PacedPacketInfo& pacing_info) { |
404 return rtp_sender_.TimeToSendPadding(bytes, pacing_info.probe_cluster_id); | 403 return rtp_sender_.TimeToSendPadding(bytes, pacing_info); |
405 } | 404 } |
406 | 405 |
407 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const { | 406 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const { |
408 return rtp_sender_.MaxPayloadSize(); | 407 return rtp_sender_.MaxPayloadSize(); |
409 } | 408 } |
410 | 409 |
411 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const { | 410 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const { |
412 return rtp_sender_.MaxRtpPacketSize(); | 411 return rtp_sender_.MaxRtpPacketSize(); |
413 } | 412 } |
414 | 413 |
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879 StreamDataCountersCallback* | 878 StreamDataCountersCallback* |
880 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 879 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
881 return rtp_sender_.GetRtpStatisticsCallback(); | 880 return rtp_sender_.GetRtpStatisticsCallback(); |
882 } | 881 } |
883 | 882 |
884 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( | 883 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( |
885 const BitrateAllocation& bitrate) { | 884 const BitrateAllocation& bitrate) { |
886 rtcp_sender_.SetVideoBitrateAllocation(bitrate); | 885 rtcp_sender_.SetVideoBitrateAllocation(bitrate); |
887 } | 886 } |
888 } // namespace webrtc | 887 } // namespace webrtc |
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