commit 43a5dd86c20df887496979acf5041f9920b1b501
author Florent Castelli <orphis@webrtc.org> Wed Apr 12 10:45:07 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Aug 24 13:18:04 2023
tree 93fad45120f542e8001c0a173b2ffa121db5ed81
parent f45c2ceae06382cdeae1f832888a480ddae26043

Implement codec selection api

The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.

Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}

api/rtp_parameters.h

diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index e277e7d..91455e2 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -520,6 +520,9 @@
   // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
   bool adaptive_ptime = false;
 
+  // Allow changing the used codec for this encoding.
+  absl::optional<RtpCodec> codec;
+
   bool operator==(const RtpEncodingParameters& o) const {
     return ssrc == o.ssrc && bitrate_priority == o.bitrate_priority &&
            network_priority == o.network_priority &&
@@ -530,7 +533,7 @@
            scale_resolution_down_by == o.scale_resolution_down_by &&
            active == o.active && rid == o.rid &&
            adaptive_ptime == o.adaptive_ptime &&
-           requested_resolution == o.requested_resolution;
+           requested_resolution == o.requested_resolution && codec == o.codec;
   }
   bool operator!=(const RtpEncodingParameters& o) const {
     return !(*this == o);