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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 2867713003: Remove hardcoded kValueSizeBytes values from variable-length header extensions. (Closed)
Patch Set: Patch 1 Created 3 years, 7 months ago
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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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410 return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms, 410 return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
411 retransmission, pacing_info); 411 retransmission, pacing_info);
412 } 412 }
413 413
414 size_t ModuleRtpRtcpImpl::TimeToSendPadding( 414 size_t ModuleRtpRtcpImpl::TimeToSendPadding(
415 size_t bytes, 415 size_t bytes,
416 const PacedPacketInfo& pacing_info) { 416 const PacedPacketInfo& pacing_info) {
417 return rtp_sender_->TimeToSendPadding(bytes, pacing_info); 417 return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
418 } 418 }
419 419
420 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const { 420 size_t ModuleRtpRtcpImpl::MaxPayloadSize(
421 return rtp_sender_->MaxPayloadSize(); 421 rtc::ArrayView<const RtpExtensionSize> extension_sizes) const {
422 return rtp_sender_->MaxPayloadSize(extension_sizes);
422 } 423 }
423 424
424 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const { 425 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
425 return rtp_sender_->MaxRtpPacketSize(); 426 return rtp_sender_->MaxRtpPacketSize();
426 } 427 }
427 428
428 void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) { 429 void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) {
429 RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE) 430 RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
430 << "rtp packet size too large: " << rtp_packet_size; 431 << "rtp packet size too large: " << rtp_packet_size;
431 RTC_DCHECK_GT(rtp_packet_size, packet_overhead_) 432 RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
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888 StreamDataCountersCallback* 889 StreamDataCountersCallback*
889 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 890 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
890 return rtp_sender_->GetRtpStatisticsCallback(); 891 return rtp_sender_->GetRtpStatisticsCallback();
891 } 892 }
892 893
893 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( 894 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
894 const BitrateAllocation& bitrate) { 895 const BitrateAllocation& bitrate) {
895 rtcp_sender_.SetVideoBitrateAllocation(bitrate); 896 rtcp_sender_.SetVideoBitrateAllocation(bitrate);
896 } 897 }
897 } // namespace webrtc 898 } // namespace webrtc
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