Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 832866a2a943f5e889ab89d9fd07295e5ad82d8d
https://github.com/WebKit/WebKit/commit/832866a2a943f5e889ab89d9fd07295e5ad82d8d
Author: David Kilzer <[email protected]>
Date: 2023-11-28 (Tue, 28 Nov 2023)
Changed paths:
M
Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
M
Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
A
Source/ThirdParty/libwebrtc/WebKit/0001-WebRTC-Release-assertion-in-webrtc-RtpPacketizerH264.patch
Log Message:
-----------
[WebRTC] Release assertion in webrtc::RtpPacketizerH264::PacketizeStapA on
bad input
https://bugs.webkit.org/show_bug.cgi?id=265422
<rdar://118859268>
Reviewed by Youenn Fablet.
Change release assertion into a runtime check to avoid a crash.
*
Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc:
(webrtc::RtpPacketizerH264::GeneratePackets):
- Check return value of PacketizeStapA() and return false if
std::nullopt was returned.
(webrtc::RtpPacketizerH264::PacketizeStapA):
- Change implementation to return std::optional<size_t>.
- Return std::nullopt instead of crashing if fragment.size() == 0.
*
Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h:
(webrtc::RtpPacketizerH264::PacketizeStapA):
- Change declaration to return std::optional<size_t>.
*
Source/ThirdParty/libwebrtc/WebKit/0001-WebRTC-Release-assertion-in-webrtc-RtpPacketizerH264.patch:
Add.
Canonical link: https://commits.webkit.org/271215@main
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