Bug 1876843 - Vendor libwebrtc from 3df661f798

Upstream commit: https://webrtc.googlesource.com/src/+/3df661f79828194d0acb51a1480833bafd9e5066
    [121] Allow RTP retransmission for cloned encoded Video Frames

    Fix the unintended disabling of RTP retransmissions for cloned encoded
    frames, caused by passing an infinite "expected_retransmission_time".
    Instead use a constant 10ms for now. For frames encoded locally, this is
    set from an estimate of the RTT, but we currently don't have access to
    that here (TODO added to pipe it through)

    If an integration is cloning and then sending frames it received, it's
    almost certainly resending received media to other peers on a local
    network, so 10ms is a fair upperbound.

    Tested locally with Chrome on Mac, configuring packet drops & observing
    on chrome://webrtc-internals that retransmission packets are now sent.

    (cherry picked from commit 3e801c32086be59e502e276ff5d6beea42069582)

    No-Try: True
    Bug: chromium:1512631
    Change-Id: I2483415dc7e0079f8a7b66f6607f4907698514c4
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331900
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Commit-Queue: Tony Herre <herre@google.com>
    Cr-Original-Commit-Position: refs/heads/main@{#41405}
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332261
    Reviewed-by: Stefan Holmer <stefan@webrtc.org>
    Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
    Reviewed-by: Henrik Boström <hbos@webrtc.org>
    Cr-Commit-Position: refs/branch-heads/6167@{#3}
    Cr-Branched-From: ece5cb83715dea85617114b6d4e981fdee2623ba-refs/heads/main@{#41315}

third_party/libwebrtc/README.moz-ff-commit

@@ -27678,3 +27678,6 @@ ece5cb8371
 # MOZ_LIBWEBRTC_SRC=/Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 9f1e1925f3
+# MOZ_LIBWEBRTC_SRC=/Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+3df661f798

third_party/libwebrtc/README.mozilla

@@ -18476,3 +18476,5 @@ libwebrtc updated from /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc c
 libwebrtc updated from /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-02-10T23:02:30.081484.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-02-10T23:04:03.473809.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /Users/jan-ivar/moz/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-02-11T17:10:57.292037.

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc

@@ -24,6 +24,12 @@
 namespace webrtc {
 namespace {
 
+// Using a reasonable default of 10ms for the retransmission delay for frames
+// not coming from this sender's encoder. This is usually taken from an
+// estimate of the RTT of the link,so 10ms should be a reasonable estimate for
+// frames being re-transmitted to a peer, probably on the same network.
+const TimeDelta kDefaultRetransmissionsTime = TimeDelta::Millis(10);
+
 class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
  public:
   TransformableVideoSenderFrame(const EncodedImage& encoded_image,
@@ -217,15 +223,17 @@ void RTPSenderVideoFrameTransformerDelegate::SendVideo(
     auto* transformed_video_frame =
         static_cast<TransformableVideoFrameInterface*>(transformed_frame.get());
     VideoFrameMetadata metadata = transformed_video_frame->Metadata();
-    sender_->SendVideo(
-        transformed_video_frame->GetPayloadType(), metadata.GetCodec(),
-        transformed_video_frame->GetTimestamp(),
-        /*capture_time=*/Timestamp::MinusInfinity(),
-        transformed_video_frame->GetData(),
-        transformed_video_frame->GetData().size(),
-        RTPVideoHeader::FromMetadata(metadata),
-        /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
-        metadata.GetCsrcs());
+    // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the
+    // retransmission time instead of a const default, in the same way as a
+    // locally encoded frame.
+    sender_->SendVideo(transformed_video_frame->GetPayloadType(),
+                       metadata.GetCodec(),
+                       transformed_video_frame->GetTimestamp(),
+                       /*capture_time=*/Timestamp::MinusInfinity(),
+                       transformed_video_frame->GetData(),
+                       transformed_video_frame->GetData().size(),
+                       RTPVideoHeader::FromMetadata(metadata),
+                       kDefaultRetransmissionsTime, metadata.GetCsrcs());
   }
 }
 
@@ -270,13 +278,14 @@ std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
                                  ? VideoFrameType::kVideoFrameKey
                                  : VideoFrameType::kVideoFrameDelta;
   // TODO(bugs.webrtc.org/14708): Fill in other EncodedImage parameters
-
+  // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the
+  // retransmission time instead of a const default, in the same way as a
+  // locally encoded frame.
   VideoFrameMetadata metadata = original->Metadata();
   RTPVideoHeader new_header = RTPVideoHeader::FromMetadata(metadata);
   return std::make_unique<TransformableVideoSenderFrame>(
       encoded_image, new_header, original->GetPayloadType(), new_header.codec,
-      original->GetTimestamp(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
+      original->GetTimestamp(), kDefaultRetransmissionsTime,
       original->GetSsrc(), metadata.GetCsrcs(), original->GetRid());
 }
 

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc

@@ -83,7 +83,7 @@ class RtpSenderVideoFrameTransformerDelegateTest : public ::testing::Test {
     delegate->TransformFrame(
         /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
         encoded_image, RTPVideoHeader::FromMetadata(metadata),
-        /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+        /*expected_retransmission_time=*/TimeDelta::Millis(10));
     return frame;
   }
 
@@ -123,7 +123,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   delegate->TransformFrame(
       /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
       encoded_image, RTPVideoHeader(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+      /*expected_retransmission_time=*/TimeDelta::Millis(10));
 }
 
 TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
@@ -260,7 +260,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
       test_sender_,
       SendVideo(payload_type, absl::make_optional(kVideoCodecVP8), timestamp,
                 /*capture_time=*/Timestamp::MinusInfinity(), buffer, _, _,
-                /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
+                /*expected_retransmission_time=*/TimeDelta::Millis(10),
                 frame_csrcs))
       .WillOnce(WithoutArgs([&] {
         event.Set();
@@ -310,7 +310,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   delegate->TransformFrame(
       /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
       encoded_image, RTPVideoHeader(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+      /*expected_retransmission_time=*/TimeDelta::Millis(10));
 }
 
 }  // namespace

third_party/libwebrtc/moz-patch-stack/0087.patch

@@ -57,10 +57,10 @@ index ebca7aaa75..ede8fdc3d6 100644
                : nullptr) {
    if (frame_transformer_delegate_)
 diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
-index 26152df0f0..561056e55b 100644
+index 2bb71941f9..ff15840529 100644
 --- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
 +++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
-@@ -32,7 +32,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
+@@ -38,7 +38,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
                                  uint32_t rtp_timestamp,
                                  TimeDelta expected_retransmission_time,
                                  uint32_t ssrc,
@@ -70,7 +70,7 @@ index 26152df0f0..561056e55b 100644
        : encoded_data_(encoded_image.GetEncodedData()),
          pre_transform_payload_size_(encoded_image.size()),
          header_(video_header),
-@@ -44,7 +45,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
+@@ -50,7 +51,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
          capture_time_identifier_(encoded_image.CaptureTimeIdentifier()),
          expected_retransmission_time_(expected_retransmission_time),
          ssrc_(ssrc),
@@ -80,7 +80,7 @@ index 26152df0f0..561056e55b 100644
      RTC_DCHECK_GE(payload_type_, 0);
      RTC_DCHECK_LE(payload_type_, 127);
    }
-@@ -107,6 +109,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
+@@ -113,6 +115,8 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
      return mime_type + CodecTypeToPayloadString(*codec_type_);
    }
 
@@ -89,15 +89,15 @@ index 26152df0f0..561056e55b 100644
   private:
    rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
    const size_t pre_transform_payload_size_;
-@@ -121,6 +125,7 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
+@@ -127,6 +131,7 @@ class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
 
    uint32_t ssrc_;
    std::vector<uint32_t> csrcs_;
 +  const std::string rid_;
  };
  }  // namespace
 
-@@ -128,10 +133,12 @@ RTPSenderVideoFrameTransformerDelegate::RTPSenderVideoFrameTransformerDelegate(
+@@ -134,10 +139,12 @@ RTPSenderVideoFrameTransformerDelegate::RTPSenderVideoFrameTransformerDelegate(
      RTPVideoFrameSenderInterface* sender,
      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
      uint32_t ssrc,
@@ -110,7 +110,7 @@ index 26152df0f0..561056e55b 100644
        transformation_queue_(task_queue_factory->CreateTaskQueue(
            "video_frame_transformer",
            TaskQueueFactory::Priority::NORMAL)) {}
-@@ -162,7 +169,7 @@ bool RTPSenderVideoFrameTransformerDelegate::TransformFrame(
+@@ -168,7 +175,7 @@ bool RTPSenderVideoFrameTransformerDelegate::TransformFrame(
    frame_transformer_->Transform(std::make_unique<TransformableVideoSenderFrame>(
        encoded_image, video_header, payload_type, codec_type, rtp_timestamp,
        expected_retransmission_time, ssrc_,
@@ -119,10 +119,10 @@ index 26152df0f0..561056e55b 100644
    return true;
  }
 
-@@ -261,7 +268,7 @@ std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
+@@ -270,7 +277,7 @@ std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
+   return std::make_unique<TransformableVideoSenderFrame>(
        encoded_image, new_header, original->GetPayloadType(), new_header.codec,
-       original->GetTimestamp(),
-       /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
+       original->GetTimestamp(), kDefaultRetransmissionsTime,
 -      original->GetSsrc(), metadata.GetCsrcs());
 +      original->GetSsrc(), metadata.GetCsrcs(), original->GetRid());
  }

third_party/libwebrtc/moz-patch-stack/0088.patch

@@ -10,7 +10,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/91d9e8b6a5c430a73
  1 file changed, 9 insertions(+)
 
 diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
-index 561056e55b..8512b21776 100644
+index ff15840529..ae9eb6b4bd 100644
 --- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
 +++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
 @@ -18,6 +18,7 @@
@@ -21,7 +21,7 @@ index 561056e55b..8512b21776 100644
  #include "rtc_base/logging.h"
 
  namespace webrtc {
-@@ -249,6 +250,14 @@ void RTPSenderVideoFrameTransformerDelegate::Reset() {
+@@ -257,6 +258,14 @@ void RTPSenderVideoFrameTransformerDelegate::Reset() {
      MutexLock lock(&sender_lock_);
      sender_ = nullptr;
    }