Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 9 insertions, 5 deletions

diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
index ac32410aed..6d3c011862 100644
--- a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
@@ -58,7 +58,8 @@ class TransformableOutgoingAudioFrame
       absl::optional<uint64_t> absolute_capture_timestamp_ms,
       uint32_t ssrc,
       std::vector<uint32_t> csrcs,
-      const std::string& codec_mime_type)
+      const std::string& codec_mime_type,
+      absl::optional<uint16_t> sequence_number)
       : frame_type_(frame_type),
         payload_type_(payload_type),
         rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
@@ -66,7 +67,8 @@ class TransformableOutgoingAudioFrame
         absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
         ssrc_(ssrc),
         csrcs_(std::move(csrcs)),
-        codec_mime_type_(codec_mime_type) {}
+        codec_mime_type_(codec_mime_type),
+        sequence_number_(sequence_number) {}
   ~TransformableOutgoingAudioFrame() override = default;
   rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
   void SetData(rtc::ArrayView<const uint8_t> data) override {
@@ -88,7 +90,7 @@ class TransformableOutgoingAudioFrame
   }
 
   const absl::optional<uint16_t> SequenceNumber() const override {
-    return absl::nullopt;
+    return sequence_number_;
   }
 
   void SetRTPTimestamp(uint32_t rtp_timestamp_with_offset) override {
@@ -108,6 +110,7 @@ class TransformableOutgoingAudioFrame
   uint32_t ssrc_;
   std::vector<uint32_t> csrcs_;
   std::string codec_mime_type_;
+  absl::optional<uint16_t> sequence_number_;
 };
 }  // namespace
 
@@ -155,7 +158,8 @@ void ChannelSendFrameTransformerDelegate::Transform(
       std::make_unique<TransformableOutgoingAudioFrame>(
           frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
           absolute_capture_timestamp_ms, ssrc,
-          /*csrcs=*/std::vector<uint32_t>(), codec_mimetype));
+          /*csrcs=*/std::vector<uint32_t>(), codec_mimetype,
+          /*sequence_number=*/absl::nullopt));
 }
 
 void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
@@ -203,7 +207,7 @@ std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
       original->GetPayloadType(), original->GetTimestamp(),
       original->GetData().data(), original->GetData().size(),
       original->AbsoluteCaptureTimestamp(), original->GetSsrc(),
-      std::move(csrcs), original->GetMimeType());
+      std::move(csrcs), original->GetMimeType(), original->SequenceNumber());
 }
 
 }  // namespace webrtc