Implement the mixer-to-client per CSRC audio level extension (RFC 6465).
This is loosely based on the similar implementation in gecko.
Bug: webrtc:9965
Change-Id: I5203a05e1c34ca6f97bd1b143790f95ff245e340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219791
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34102}
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index 7fe9f2b..3ecaaf8 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -357,6 +357,11 @@
static constexpr char kVideoFrameTrackingIdUri[] =
"http://www.webrtc.org/experiments/rtp-hdrext/video-frame-tracking-id";
+ // Header extension for Mixer-to-Client audio levels per CSRC as defined in
+ // https://tools.ietf.org/html/rfc6465
+ static constexpr char kCsrcAudioLevelsUri[] =
+ "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+
// Inclusive min and max IDs for two-byte header extensions and one-byte
// header extensions, per RFC8285 Section 4.2-4.3.
static constexpr int kMinId = 1;