tor-browser

commit c797b290fa2d3da3d72e3e6f16a7730a2d418155
parent 52b87899ccd6eea88225ddced040b15683bf1446
Author: Dan Baker <dbaker@mozilla.com>
Date:   Thu, 23 Oct 2025 14:35:11 -0600

Bug 1995393 - Vendor libwebrtc from 53beafc5f7

Upstream commit: https://webrtc.googlesource.com/src/+/53beafc5f7d6e3954f15426aecc79a8e9c663a13
    Fix RtpFileSource when reading full RTP packets

    Bug: webrtc:437816262
    Change-Id: If69e649a985600d91ca8cece1ca83d82dfeda6c3
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404040
    Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
    Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#45321}

Diffstat:
M
third_party/libwebrtc/README.mozilla.last-vendor
 | 
4
++--
M
third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
 | 
4
+++-

2 files changed, 5 insertions(+), 3 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T20:32:34.070371+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T20:34:54.025863+00:00.
 # base of lastest vendoring
-3ced074b14
+53beafc5f7
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -75,7 +75,9 @@ std::unique_ptr<RtpPacketReceived> RtpFileSource::NextPacket() {
       continue;
     }
     rtp_packet->set_arrival_time(Timestamp::Millis(temp_packet.time_ms));
-    if (temp_packet.original_length > rtp_packet->headers_size()) {
+
+    // Simulate payload if only the RTP header was written in the file.
+    if (temp_packet.original_length > rtp_packet->size()) {
       size_t payload_size =
           temp_packet.original_length - rtp_packet->headers_size();
       if (rtp_packet->has_padding()) {