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Updated comments for RtpEncodingParameters.

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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}
This commit is contained in:
Seth Hampson 2018-02-12 14:14:39 -08:00 committed by Commit Bot
parent 73f29cbcc1
commit a881ac0ec9
1 changed files with 22 additions and 7 deletions
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29
api/rtpparameters.h
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@ -362,8 +362,12 @@ struct RtpEncodingParameters {
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 @@ struct RtpEncodingParameters {
// 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 @@ struct RtpEncodingParameters {
// 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.