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_; }