From 26617bef59b6efed61472f05abb3a5a59c76cf1c Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 23 Jan 2025 18:24:56 +0100 Subject: [PATCH] Make AV1 even payload size default-on when packetizer is used directly MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 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 Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#43807} --- modules/rtp_rtcp/source/rtp_format.h | 4 ++-- modules/rtp_rtcp/source/rtp_packetizer_av1.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h index 097481d24e..95d4c60d1f 100644 --- a/modules/rtp_rtcp/source/rtp_format.h +++ b/modules/rtp_rtcp/source/rtp_format.h @@ -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; diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1.h b/modules/rtp_rtcp/source/rtp_packetizer_av1.h index 79f9b323d2..b28fa61562 100644 --- a/modules/rtp_rtcp/source/rtp_packetizer_av1.h +++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.h @@ -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_; }