tor-browser

commit b075be07dcce49956931fc699fe517a72fbe3eb4
parent 0f0bdd4214264bc7f29b286874ef3cbd278a5a9c
Author: Dan Baker <dbaker@mozilla.com>
Date:   Mon, 27 Oct 2025 14:53:13 -0600

Bug 1995393 - Vendor libwebrtc from 01e5a0d33f

Upstream commit: https://webrtc.googlesource.com/src/+/01e5a0d33f639e120861352093eaea06e8c9f29d
    Make WebRTC-RTP-Lifetime Unified Plan-only.

    We accidentally only gated the outbound-rtp lifetime on Unified Plan
    but not the inbound-rtp change. This CL fixes that such that when we
    ship "WebRTC-RTP-Lifetime" this does not impact Plan B.

    Bug: chromium:406585888
    Change-Id: I2e058824bf4695bca568508441e2ad51b4a828a6
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406725
    Commit-Queue: Harald Alvestrand <hta@webrtc.org>
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Auto-Submit: Henrik Boström <hbos@webrtc.org>
    Commit-Queue: Henrik Boström <hbos@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#45473}

Diffstat:
M
third_party/libwebrtc/README.mozilla.last-vendor
 | 
4
++--
M
third_party/libwebrtc/pc/rtc_stats_collector.cc
 | 
9
++++-----

2 files changed, 6 insertions(+), 7 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 2025-10-27T20:50:36.608619+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T20:53:00.338250+00:00.
 # base of lastest vendoring
-b2a3a643c4
+01e5a0d33f
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.cc b/third_party/libwebrtc/pc/rtc_stats_collector.cc
@@ -1718,7 +1718,8 @@ void RTCStatsCollector::ProduceRTPStreamStats_n(
   RTC_DCHECK_RUN_ON(network_thread_);
   Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
-  bool spec_lifetime = env_.field_trials().IsEnabled("WebRTC-RTP-Lifetime");
+  bool spec_lifetime =
+      is_unified_plan_ && env_.field_trials().IsEnabled("WebRTC-RTP-Lifetime");
   for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) {
     if (stats.media_type == MediaType::AUDIO) {
       ProduceAudioRTPStreamStats_n(timestamp, stats, spec_lifetime, report);
@@ -1805,8 +1806,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
     if (!voice_sender_info.connected()) {
       continue;  // The SSRC is not known yet.
     }
-    if (spec_lifetime && is_unified_plan_ &&
-        !stats.current_direction.has_value()) {
+    if (spec_lifetime && !stats.current_direction.has_value()) {
       continue;  // The SSRC is known but the O/A has not completed.
     }
     auto outbound_audio = CreateOutboundRTPStreamStatsFromVoiceSenderInfo(
@@ -1912,8 +1912,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
     if (!video_sender_info.connected()) {
       continue;  // The SSRC is not known yet.
     }
-    if (spec_lifetime && is_unified_plan_ &&
-        !stats.current_direction.has_value()) {
+    if (spec_lifetime && !stats.current_direction.has_value()) {
       continue;  // The SSRC is known but the O/A has not completed.
     }
     auto outbound_video = CreateOutboundRTPStreamStatsFromVideoSenderInfo(