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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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