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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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602 return 0; | 602 return 0; |
603 } | 603 } |
604 | 604 |
605 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { | 605 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { |
606 // The max size of the nack list should be large enough to accommodate the | 606 // The max size of the nack list should be large enough to accommodate the |
607 // the number of packets (frames) resulting from the increased delay. | 607 // the number of packets (frames) resulting from the increased delay. |
608 // Roughly estimating for ~40 packets per frame @ 30fps. | 608 // Roughly estimating for ~40 packets per frame @ 30fps. |
609 return target_delay_ms * 40 * 30 / 1000; | 609 return target_delay_ms * 40 * 30 / 1000; |
610 } | 610 } |
611 | 611 |
612 void ViEChannel::EnableRemb(bool enable) { | |
613 rtp_rtcp_modules_[0]->SetREMBStatus(enable); | |
614 } | |
615 | |
616 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { | 612 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { |
617 // Disable any previous registrations of this extension to avoid errors. | 613 // Disable any previous registrations of this extension to avoid errors. |
618 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 614 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
619 rtp_rtcp->DeregisterSendRtpHeaderExtension( | 615 rtp_rtcp->DeregisterSendRtpHeaderExtension( |
620 kRtpExtensionTransmissionTimeOffset); | 616 kRtpExtensionTransmissionTimeOffset); |
621 } | 617 } |
622 if (!enable) | 618 if (!enable) |
623 return 0; | 619 return 0; |
624 // Enable the extension. | 620 // Enable the extension. |
625 int error = 0; | 621 int error = 0; |
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1248 CriticalSectionScoped cs(crit_.get()); | 1244 CriticalSectionScoped cs(crit_.get()); |
1249 receive_stats_callback_ = receive_statistics_proxy; | 1245 receive_stats_callback_ = receive_statistics_proxy; |
1250 } | 1246 } |
1251 | 1247 |
1252 void ViEChannel::SetIncomingVideoStream( | 1248 void ViEChannel::SetIncomingVideoStream( |
1253 IncomingVideoStream* incoming_video_stream) { | 1249 IncomingVideoStream* incoming_video_stream) { |
1254 CriticalSectionScoped cs(crit_.get()); | 1250 CriticalSectionScoped cs(crit_.get()); |
1255 incoming_video_stream_ = incoming_video_stream; | 1251 incoming_video_stream_ = incoming_video_stream; |
1256 } | 1252 } |
1257 } // namespace webrtc | 1253 } // namespace webrtc |
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