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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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770 } | 770 } |
771 | 771 |
772 // Send a TelephoneEvent tone using RFC 2833 (4733). | 772 // Send a TelephoneEvent tone using RFC 2833 (4733). |
773 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( | 773 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( |
774 const uint8_t key, | 774 const uint8_t key, |
775 const uint16_t time_ms, | 775 const uint16_t time_ms, |
776 const uint8_t level) { | 776 const uint8_t level) { |
777 return rtp_sender_.SendTelephoneEvent(key, time_ms, level); | 777 return rtp_sender_.SendTelephoneEvent(key, time_ms, level); |
778 } | 778 } |
779 | 779 |
780 // Set audio packet size, used to determine when it's time to send a DTMF | |
781 // packet in silence (CNG). | |
782 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( | 780 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( |
783 const uint16_t packet_size_samples) { | 781 const uint16_t packet_size_samples) { |
784 return rtp_sender_.SetAudioPacketSize(packet_size_samples); | 782 return audio_ ? 0 : -1; |
785 } | 783 } |
786 | 784 |
787 int32_t ModuleRtpRtcpImpl::SetAudioLevel( | 785 int32_t ModuleRtpRtcpImpl::SetAudioLevel( |
788 const uint8_t level_d_bov) { | 786 const uint8_t level_d_bov) { |
789 return rtp_sender_.SetAudioLevel(level_d_bov); | 787 return rtp_sender_.SetAudioLevel(level_d_bov); |
790 } | 788 } |
791 | 789 |
792 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( | 790 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( |
793 const KeyFrameRequestMethod method) { | 791 const KeyFrameRequestMethod method) { |
794 key_frame_req_method_ = method; | 792 key_frame_req_method_ = method; |
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947 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( | 945 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( |
948 StreamDataCountersCallback* callback) { | 946 StreamDataCountersCallback* callback) { |
949 rtp_sender_.RegisterRtpStatisticsCallback(callback); | 947 rtp_sender_.RegisterRtpStatisticsCallback(callback); |
950 } | 948 } |
951 | 949 |
952 StreamDataCountersCallback* | 950 StreamDataCountersCallback* |
953 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 951 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
954 return rtp_sender_.GetRtpStatisticsCallback(); | 952 return rtp_sender_.GetRtpStatisticsCallback(); |
955 } | 953 } |
956 } // namespace webrtc | 954 } // namespace webrtc |
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