Updated comments for RtpEncodingParameters.
Currently with the RtpEncodingParameters the active field is supported
per simulcast layer, but max_bitrate_bps and bitrate_priority are just
supoorted per rtp sender. Updated the comments to make this more clear
and added TODOs with bugs.
Bug: webrtc:8819
Change-Id: I130f6dda0896079b5082af58a2693b898d6e22f0
Reviewed-on: https://webrtc-review.googlesource.com/52141
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22007}diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index 00ae094..d9ac1b6 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -362,8 +362,12 @@
rtc::Optional<DtxStatus> dtx;
// The relative bitrate priority of this encoding. Currently this is
- // implemented on the sender level (using the first RtpEncodingParameters
- // of the rtp parameters).
+ // implemented for the entire rtp sender by using the value of the first
+ // encoding parameter.
+ // TODO(webrtc.bugs.org/8630): Implement this per encoding parameter.
+ // Currently there is logic for how bitrate is distributed per simulcast layer
+ // in the VideoBitrateAllocator. This must be updated to incorporate relative
+ // bitrate priority.
double bitrate_priority = kDefaultBitratePriority;
// Indicates the preferred duration of media represented by a packet in
@@ -376,7 +380,16 @@
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
- // bitrate.
+ // bitrate. Currently this is implemented for the entire rtp sender by using
+ // the value of the first encoding parameter.
+ //
+ // TODO(webrtc.bugs.org/8655): Implement this per encoding parameter.
+ // Current implementation for a sender:
+ // The max bitrate is decided by taking the minimum of the first encoding
+ // parameter's max_bitrate_bps and the max bitrate specified by the sdp with
+ // the b=AS attribute. In the case of simulcast video, default values are used
+ // for each simulcast layer, and if there is some bitrate left over from the
+ // sender's max bitrate then it will roll over into the highest quality layer.
//
// Just called "maxBitrate" in ORTC spec.
//
@@ -397,10 +410,12 @@
// TODO(deadbeef): Not implemented.
double scale_framerate_down_by = 1.0;
- // For an RtpSender, set to true to cause this encoding to be sent, and false
- // for it not to be sent.
- // TODO(bugs.webrtc.org/8653): Currently this is implemented per sender.
- // Implement per-encoding.
+ // For an RtpSender, set to true to cause this encoding to be encoded and
+ // sent, and false for it not to be encoded and sent. This allows control
+ // across multiple encodings of a sender for turning simulcast layers on and
+ // off.
+ // TODO(webrtc.bugs.org/8807): Updating this parameter will trigger an encoder
+ // reset, but this isn't necessarily required.
bool active = true;
// Value to use for RID RTP header extension.
