Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(49)

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

Issue 9188022: Initialized a variable to fix -Wmaybe-uninitialized warning. (Closed) Base URL: http://git.chromium.org/external/webrtc/src.git@master
Patch Set: Created 8 years, 11 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
« no previous file with comments | « no previous file | no next file » | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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 704 matching lines...) Expand 10 before | Expand all | Expand 10 after
715 const RTPVideoTypeHeader* rtpTypeHdr) 715 const RTPVideoTypeHeader* rtpTypeHdr)
716 { 716 {
717 { 717 {
718 // Drop this packet if we're not sending media packets 718 // Drop this packet if we're not sending media packets
719 CriticalSectionScoped cs(_sendCritsect); 719 CriticalSectionScoped cs(_sendCritsect);
720 if (!_sendingMedia) 720 if (!_sendingMedia)
721 { 721 {
722 return 0; 722 return 0;
723 } 723 }
724 } 724 }
725 RtpVideoCodecTypes videoType; 725 RtpVideoCodecTypes videoType = kRtpNoVideo;
726 if(CheckPayloadType(payloadType, videoType) != 0) 726 if(CheckPayloadType(payloadType, videoType) != 0)
727 { 727 {
728 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument faile d to find payloadType:%d", __FUNCTION__, payloadType); 728 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument faile d to find payloadType:%d", __FUNCTION__, payloadType);
729 return -1; 729 return -1;
730 } 730 }
731 // update keepalive so that we don't trigger keepalive messages while sendin g data 731 // update keepalive so that we don't trigger keepalive messages while sendin g data
732 _keepAliveLastSent = _clock.GetTimeInMS(); 732 _keepAliveLastSent = _clock.GetTimeInMS();
733 733
734 if(_audioConfigured) 734 if(_audioConfigured)
735 { 735 {
(...skipping 1006 matching lines...) Expand 10 before | Expand all | Expand 10 after
1742 1742
1743 { 1743 {
1744 if(_audioConfigured) 1744 if(_audioConfigured)
1745 { 1745 {
1746 return -1; 1746 return -1;
1747 } 1747 }
1748 return _video->SetFECUepProtection(keyUseUepProtection, 1748 return _video->SetFECUepProtection(keyUseUepProtection,
1749 deltaUseUepProtection); 1749 deltaUseUepProtection);
1750 } 1750 }
1751 } // namespace webrtc 1751 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698