Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(223)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

Issue 1677003002: Fix null-pointer dereference in RTPSenderVideo. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rename video_header Created 4 years, 10 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 | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | 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) 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 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
45 const uint32_t maxBitRate); 45 const uint32_t maxBitRate);
46 46
47 int32_t SendVideo(const RtpVideoCodecTypes videoType, 47 int32_t SendVideo(const RtpVideoCodecTypes videoType,
48 const FrameType frameType, 48 const FrameType frameType,
49 const int8_t payloadType, 49 const int8_t payloadType,
50 const uint32_t captureTimeStamp, 50 const uint32_t captureTimeStamp,
51 int64_t capture_time_ms, 51 int64_t capture_time_ms,
52 const uint8_t* payloadData, 52 const uint8_t* payloadData,
53 const size_t payloadSize, 53 const size_t payloadSize,
54 const RTPFragmentationHeader* fragmentation, 54 const RTPFragmentationHeader* fragmentation,
55 const RTPVideoHeader* rtpHdr); 55 const RTPVideoHeader* video_header);
56 56
57 int32_t SendRTPIntraRequest(); 57 int32_t SendRTPIntraRequest();
58 58
59 void SetVideoCodecType(RtpVideoCodecTypes type); 59 void SetVideoCodecType(RtpVideoCodecTypes type);
60 60
61 void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate); 61 void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate);
62 62
63 uint32_t MaxConfiguredBitrateVideo() const; 63 uint32_t MaxConfiguredBitrateVideo() const;
64 64
65 // FEC 65 // FEC
(...skipping 54 matching lines...) Expand 10 before | Expand all | Expand 10 after
120 120
121 // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets 121 // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
122 // and any padding overhead. 122 // and any padding overhead.
123 Bitrate _fecOverheadRate; 123 Bitrate _fecOverheadRate;
124 // Bitrate used for video payload and RTP headers 124 // Bitrate used for video payload and RTP headers
125 Bitrate _videoBitrate; 125 Bitrate _videoBitrate;
126 }; 126 };
127 } // namespace webrtc 127 } // namespace webrtc
128 128
129 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ 129 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698