Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(250)

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

Issue 1226143013: Merge methods for configuring NACK/FEC/hybrid. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 1718 matching lines...) Expand 10 before | Expand all | Expand 10 after
1729 return video_->MaxConfiguredBitrateVideo(); 1729 return video_->MaxConfiguredBitrateVideo();
1730 } 1730 }
1731 1731
1732 int32_t RTPSender::SendRTPIntraRequest() { 1732 int32_t RTPSender::SendRTPIntraRequest() {
1733 if (audio_configured_) { 1733 if (audio_configured_) {
1734 return -1; 1734 return -1;
1735 } 1735 }
1736 return video_->SendRTPIntraRequest(); 1736 return video_->SendRTPIntraRequest();
1737 } 1737 }
1738 1738
1739 int32_t RTPSender::SetGenericFECStatus(bool enable, 1739 void RTPSender::SetGenericFECStatus(bool enable,
1740 uint8_t payload_type_red, 1740 uint8_t payload_type_red,
1741 uint8_t payload_type_fec) { 1741 uint8_t payload_type_fec) {
1742 if (audio_configured_) { 1742 DCHECK(!audio_configured_);
1743 return -1;
1744 }
1745 video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec); 1743 video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
1746 return 0;
1747 } 1744 }
1748 1745
1749 int32_t RTPSender::GenericFECStatus(bool* enable, 1746 void RTPSender::GenericFECStatus(bool* enable,
1750 uint8_t* payload_type_red, 1747 uint8_t* payload_type_red,
1751 uint8_t* payload_type_fec) const { 1748 uint8_t* payload_type_fec) const {
1752 if (audio_configured_) { 1749 DCHECK(!audio_configured_);
1753 return -1;
1754 }
1755 video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec); 1750 video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
1756 return 0;
1757 } 1751 }
1758 1752
1759 int32_t RTPSender::SetFecParameters( 1753 int32_t RTPSender::SetFecParameters(
1760 const FecProtectionParams *delta_params, 1754 const FecProtectionParams *delta_params,
1761 const FecProtectionParams *key_params) { 1755 const FecProtectionParams *key_params) {
1762 if (audio_configured_) { 1756 if (audio_configured_) {
1763 return -1; 1757 return -1;
1764 } 1758 }
1765 video_->SetFecParameters(delta_params, key_params); 1759 video_->SetFecParameters(delta_params, key_params);
1766 return 0; 1760 return 0;
(...skipping 88 matching lines...) Expand 10 before | Expand all | Expand 10 after
1855 CriticalSectionScoped lock(send_critsect_.get()); 1849 CriticalSectionScoped lock(send_critsect_.get());
1856 1850
1857 RtpState state; 1851 RtpState state;
1858 state.sequence_number = sequence_number_rtx_; 1852 state.sequence_number = sequence_number_rtx_;
1859 state.start_timestamp = start_timestamp_; 1853 state.start_timestamp = start_timestamp_;
1860 1854
1861 return state; 1855 return state;
1862 } 1856 }
1863 1857
1864 } // namespace webrtc 1858 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender.h ('k') | webrtc/modules/video_coding/main/interface/video_coding_defines.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698