tor-browser

commit 3b23e19181d041fb567075556d46fe32b4b34663
parent 7bec92371a14caa984eda7ff379f8c8d321aad00
Author: Dan Baker <dbaker@mozilla.com>
Date:   Mon, 27 Oct 2025 14:20:37 -0600

Bug 1995393 - Vendor libwebrtc from 0c5b0d6dee

Upstream commit: https://webrtc.googlesource.com/src/+/0c5b0d6deeab042b4c773367bbcc11b746cdab82
    psnr: tweak sampling rate to make a new measurement every second more likely

    When getStats is polled at exactly 1hz the current frame sampling may
    occasionally yield no increase in measurements. Avoid this by doing a
    simple linear extrapolation using the previous and current frame and
    comparing that extrapolation to the 90k frequency.

    Bug: webrtc:375021
    Change-Id: I252a2b3fef87cbd728e40f83e7efe18f05aafc37
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406143
    Reviewed-by: Erik Språng <sprang@webrtc.org>
    Reviewed-by: Henrik Boström <hbos@webrtc.org>
    Commit-Queue: Philipp Hancke <phancke@meta.com>
    Cr-Commit-Position: refs/heads/main@{#45458}

Diffstat:
M
third_party/libwebrtc/README.mozilla.last-vendor
 | 
4
++--
M
third_party/libwebrtc/modules/video_coding/utility/frame_sampler.cc
 | 
25
+++++++++++++++++++++----
M
third_party/libwebrtc/modules/video_coding/utility/frame_sampler.h
 | 
1
+
M
third_party/libwebrtc/modules/video_coding/utility/frame_sampler_unittest.cc
 | 
19
+++++++++++++++++--

4 files changed, 41 insertions(+), 8 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 2025-10-27T20:18:26.782165+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T20:20:25.844512+00:00.
 # base of lastest vendoring
-02a85ad9fb
+0c5b0d6dee
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_sampler.cc b/third_party/libwebrtc/modules/video_coding/utility/frame_sampler.cc
@@ -10,18 +10,35 @@
 
 #include "modules/video_coding/utility/frame_sampler.h"
 
+#include <cstdint>
+
 #include "api/video/video_frame.h"
 #include "modules/include/module_common_types_public.h"
 
 namespace webrtc {
 
 constexpr int kTimestampDifference =
-    90'000 - 1;  // Sample every 90khz or once per second.
+    90'000;  // Sample every 90khz or once per second.
 
 bool FrameSampler::ShouldBeSampled(const VideoFrame& frame) {
-  if (!last_rtp_timestamp_sampled_.has_value() ||
-      (IsNewerTimestamp(frame.rtp_timestamp(),
-                        *last_rtp_timestamp_sampled_ + kTimestampDifference))) {
+  if (!last_rtp_timestamp_sampled_) {
+    // Since we can not know the frame rate from the first frame,
+    // assume 30fps for the extrapolation.
+    last_rtp_timestamp_ =
+        frame.rtp_timestamp() + kTimestampDifference / /*fps=*/30;
+    last_rtp_timestamp_sampled_ = frame.rtp_timestamp();
+    return true;
+  }
+  // Since getStats is commonly called once per second, sample if the
+  // extrapolated RTP timestamp of the next frame would be be too late for this.
+  // This is not strictly necessary but makes plotting the values once per
+  // second much easier.
+  uint32_t extrapolated_rtp_timestamp =
+      frame.rtp_timestamp() + (frame.rtp_timestamp() - *last_rtp_timestamp_);
+  last_rtp_timestamp_ = frame.rtp_timestamp();
+
+  if (IsNewerTimestamp(extrapolated_rtp_timestamp,
+                       *last_rtp_timestamp_sampled_ + kTimestampDifference)) {
     last_rtp_timestamp_sampled_ = frame.rtp_timestamp();
     return true;
   }
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_sampler.h b/third_party/libwebrtc/modules/video_coding/utility/frame_sampler.h
@@ -32,6 +32,7 @@ class FrameSampler {
 
  private:
   std::optional<uint32_t> last_rtp_timestamp_sampled_;
+  std::optional<uint32_t> last_rtp_timestamp_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_sampler_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/frame_sampler_unittest.cc
@@ -28,7 +28,22 @@ TEST(FrameSampler, SamplesBasedOnRtpTimestamp) {
   EXPECT_TRUE(sampler.ShouldBeSampled(frame));
   frame.set_rtp_timestamp(45'000);
   EXPECT_FALSE(sampler.ShouldBeSampled(frame));
-  frame.set_rtp_timestamp(90'000);
+  frame.set_rtp_timestamp(90'000 - 3'000);
+  EXPECT_TRUE(sampler.ShouldBeSampled(frame));
+}
+
+TEST(FrameSampler, SamplesBasedOnRtpTimestampDeltaLessThanOneSecond) {
+  FrameSampler sampler;
+
+  auto buffer = make_ref_counted<I420Buffer>(320, 240);
+  VideoFrame frame =
+      VideoFrame::Builder().set_video_frame_buffer(buffer).build();
+
+  frame.set_rtp_timestamp(0);
+  EXPECT_TRUE(sampler.ShouldBeSampled(frame));
+  frame.set_rtp_timestamp(3'000);
+  EXPECT_FALSE(sampler.ShouldBeSampled(frame));
+  frame.set_rtp_timestamp(90'000 - 3'000);
   EXPECT_TRUE(sampler.ShouldBeSampled(frame));
 }
 
@@ -40,7 +55,7 @@ TEST(FrameSampler, RtpTimestampWraparound) {
       VideoFrame::Builder().set_video_frame_buffer(buffer).build();
 
   // RTP timestamp wraps at 2**32.
-  frame.set_rtp_timestamp(0xffff'ffff - 4000);
+  frame.set_rtp_timestamp(0xffff'ffff - 3'000);
   EXPECT_TRUE(sampler.ShouldBeSampled(frame));
   frame.set_rtp_timestamp(41'000);
   EXPECT_FALSE(sampler.ShouldBeSampled(frame));