Make AV1 even payload size default-on when packetizer is used directly
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}
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index 097481d..95d4c60 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -41,8 +41,8 @@
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 79f9b32..b28fa61 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1.h
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.h
@@ -28,7 +28,7 @@
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_; }