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Add note about two-byte extension to VLA docs

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since the extension can be too large to fit the 16 bytes available
to one-byte extensions
  https://www.rfc-editor.org/rfc/rfc8285#section-4.2
when including the width and height fields.
Also document when those fields are sent.

BUG=webrtc:12000

Change-Id: If17f57d40c0bde9b060f223c548e407d6c124b82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40910}
This commit is contained in:
Philipp Hancke 2023-10-11 12:03:56 +03:00 committed by WebRTC LUCI CQ
parent 1a8d5292c2
commit dde1cb6212
2 changed files with 12 additions and 6 deletions
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12
docs/native-code/rtp-hdrext/video-layers-allocation00/README.md
View File

@ -22,6 +22,10 @@ rtp stream (SVC), or independent spatial layers sent on multiple rtp streams
## RTP header extension format
Note: when including the optional width, height and maximum framerate
fields, the total data length of the extension can exceed 16 bytes
and is sent as a two-byte header extension [1]
### Data layout
```
@ -68,7 +72,7 @@ alignment.
layers. Values are stored in ascending order of spatial id. Zero-padded to byte
alignment.
Target bitrate in kbps. Values are stored using leb128 encoding [1]. One value per
Target bitrate in kbps. Values are stored using leb128 encoding [2]. One value per
temporal layer. Values are stored in (RTP stream id, spatial id, temporal id)
ascending order. All bitrates are total required bitrate to receive the
corresponding layer, i.e. in simulcast mode they include only corresponding
@ -78,9 +82,11 @@ temporal layers are also included.
Resolution and framerate. Optional. Presence is inferred from the rtp header
extension size. Encoded (width - 1), 16-bit, (height - 1), 16-bit, max frame
rate 8-bit per spatial layer per RTP stream. Values are stored in (RTP stream
id, spatial id) ascending order.
id, spatial id) ascending order. Only sent when the resolution differs from the
last values, the framerate changed by more than 5fps and on key frames.
An empty layer allocation (i.e nothing sent on ssrc) is encoded as
special case with a single 0 byte.
[1] https://aomediacodec.github.io/av1-spec/#leb128
[1] https://www.rfc-editor.org/rfc/rfc8285#section-4.3
[2] https://aomediacodec.github.io/av1-spec/#leb128

6
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -118,7 +118,7 @@ absl::optional<VideoPlayoutDelay> LoadVideoPlayoutDelayOverride(
// Some packets can be skipped and the stream can still be decoded. Those
// packets are less likely to be retransmitted if they are lost.
bool PacketWillLikelyBeRequestedForRestransmitionIfLost(
bool PacketWillLikelyBeRequestedForRestransmissionIfLost(
const RTPVideoHeader& video_header) {
return IsBaseLayer(video_header) &&
!(video_header.generic.has_value()
@ -442,7 +442,7 @@ void RTPSenderVideo::AddRtpHeaderExtensions(const RTPVideoHeader& video_header,
first_packet &&
send_allocation_ != SendVideoLayersAllocation::kDontSend &&
(video_header.frame_type == VideoFrameType::kVideoFrameKey ||
PacketWillLikelyBeRequestedForRestransmitionIfLost(video_header))) {
PacketWillLikelyBeRequestedForRestransmissionIfLost(video_header))) {
VideoLayersAllocation allocation = allocation_.value();
allocation.resolution_and_frame_rate_is_valid =
send_allocation_ == SendVideoLayersAllocation::kSendWithResolution;
@ -733,7 +733,7 @@ bool RTPSenderVideo::SendVideo(int payload_type,
}
if (video_header.frame_type == VideoFrameType::kVideoFrameKey ||
PacketWillLikelyBeRequestedForRestransmitionIfLost(video_header)) {
PacketWillLikelyBeRequestedForRestransmissionIfLost(video_header)) {
// This frame will likely be delivered, no need to populate playout
// delay extensions until it changes again.
playout_delay_pending_ = false;