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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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779 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( | 779 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( |
780 const uint16_t packet_size_samples) { | 780 const uint16_t packet_size_samples) { |
781 return rtp_sender_.SetAudioPacketSize(packet_size_samples); | 781 return rtp_sender_.SetAudioPacketSize(packet_size_samples); |
782 } | 782 } |
783 | 783 |
784 int32_t ModuleRtpRtcpImpl::SetAudioLevel( | 784 int32_t ModuleRtpRtcpImpl::SetAudioLevel( |
785 const uint8_t level_d_bov) { | 785 const uint8_t level_d_bov) { |
786 return rtp_sender_.SetAudioLevel(level_d_bov); | 786 return rtp_sender_.SetAudioLevel(level_d_bov); |
787 } | 787 } |
788 | 788 |
789 // Set payload type for Redundant Audio Data RFC 2198. | |
790 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType( | |
791 const int8_t payload_type) { | |
792 return rtp_sender_.SetRED(payload_type); | |
793 } | |
794 | |
795 // Get payload type for Redundant Audio Data RFC 2198. | |
796 int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const { | |
797 return rtp_sender_.RED(payload_type); | |
798 } | |
799 | |
800 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( | 789 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( |
801 const KeyFrameRequestMethod method) { | 790 const KeyFrameRequestMethod method) { |
802 key_frame_req_method_ = method; | 791 key_frame_req_method_ = method; |
803 return 0; | 792 return 0; |
804 } | 793 } |
805 | 794 |
806 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { | 795 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { |
807 switch (key_frame_req_method_) { | 796 switch (key_frame_req_method_) { |
808 case kKeyFrameReqPliRtcp: | 797 case kKeyFrameReqPliRtcp: |
809 return SendRTCP(kRtcpPli); | 798 return SendRTCP(kRtcpPli); |
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969 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( | 958 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( |
970 StreamDataCountersCallback* callback) { | 959 StreamDataCountersCallback* callback) { |
971 rtp_sender_.RegisterRtpStatisticsCallback(callback); | 960 rtp_sender_.RegisterRtpStatisticsCallback(callback); |
972 } | 961 } |
973 | 962 |
974 StreamDataCountersCallback* | 963 StreamDataCountersCallback* |
975 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 964 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
976 return rtp_sender_.GetRtpStatisticsCallback(); | 965 return rtp_sender_.GetRtpStatisticsCallback(); |
977 } | 966 } |
978 } // namespace webrtc | 967 } // namespace webrtc |
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