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Make AV1 even payload size default-on when packetizer is used directly

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This flip default behavior for webrtc users that create packetizers without help of RtpSenderVideo class.

Bug: webrtc:42226301
Change-Id: I42fe696039334672b7d0b0ed1f87a52c3f6bf5ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43807}
This commit is contained in:
Danil Chapovalov 2025-01-23 18:24:56 +01:00 committed by WebRTC LUCI CQ
parent 13170bd177
commit 26617bef59
2 changed files with 3 additions and 3 deletions
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4
modules/rtp_rtcp/source/rtp_format.h
View File

@ -41,8 +41,8 @@ class RtpPacketizer {
PayloadSizeLimits limits,
// Codec-specific details.
const RTPVideoHeader& rtp_video_header,
// TODO(bugs.webrtc.org/15927): remove after rollout.
bool enable_av1_even_split = false);
// TODO: bugs.webrtc.org/42226301 - remove after rollout.
bool enable_av1_even_split = true);
virtual ~RtpPacketizer() = default;

2
modules/rtp_rtcp/source/rtp_packetizer_av1.h
View File

@ -28,7 +28,7 @@ class RtpPacketizerAv1 : public RtpPacketizer {
PayloadSizeLimits limits,
VideoFrameType frame_type,
bool is_last_frame_in_picture,
bool even_distribution);
bool even_distribution = true);
~RtpPacketizerAv1() override = default;
size_t NumPackets() const override { return packets_.size() - packet_index_; }