which needs to be added to the remote codecs a=fmtp:
This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.
This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.
BUG=webrtc:10107
Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
kDefaultQpMax=56 was defined in multiple places. Move it to media_constants and split it into two: VPx/AV1 and H26x values. H26x value is set to 51 which is the max bitstream QP value for H264/5.
This CL is expected to be a no-op because:
1. VideoCodec::qpMax value has not changed for VP8/9 and AV1.
2. VideoCodec::qpMax is currently not used by OpenH264 wrapper (wiring it up is out-of-scope of this CL).
3. Previous default qpMax=56 exceeded the max value for H26x (=51). External HW H26x encoders likely clamped it and used 51.
Bug: webrtc:14852
Change-Id: I1d795e695dac5c78e86ed829b24281e61066f668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324282
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40997}
Assuming 15Mbps video bitrate at 30fps, a single frame is 62500 bytes.
Add to that some fluctuations in encoder output rate and capture fps,
and frames can easily become larger than 64kb.
Given enough bandwidth and the bursty pacer, it will not be uncommon to
send the entire frame in one batch - and if the send buffer is at 64kb
then you will likely get packetloss already in the IPC packet socket,
even before the packet has reached the network card!
It's not entirely clear what the optimal size is, but given that the
receive buffer size was increased from 64kb to 256kb for high bandwidth
receive scenarios and had negligible negative effects I think it's
pretty safe to bump the send buffer to match.
There is a field trial available that can be used as circuit breaker
in case things turn south: WebRTC-SendBufferSizeBytes
Bug: webrtc:14780
Change-Id: I6c786d993181a882e6dce832ff56dc92d2a8a341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290985
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39942}
following the removal of ISAC from the code base.
BUG=webrtc:14450
Change-Id: I6faab5391bf0ef563c5dcce0bd5d8a653a87d9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294523
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39378}
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.
Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
along the lines of RTX handling but with limited support for missing
fmtp lines because of video/red.
BUG=webrtc:13178
Change-Id: Ia866c0e857da6da2ef1e4b81b51f90f534c7bb83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231948
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35107}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
provides an implementation of the rtx-time parameter from
https://tools.ietf.org/html/rfc4588#section-8
that determines the maximum time a receiver waits for a frame
before sending a PLI.
BUG=webrtc:12420
Change-Id: Iff20d92c806989cd4d56fe330d105b3dd127ed24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33627}
removes the unused
x-google-port
x-google-max-message-size
which were probably sctp-related and variants of max-message-size
and sctp-port parameters. Similarly for the
protocol
streams
parameters which seem to have been fmtp parameters.
BUG=chromium:943975
Change-Id: I21b543f717d6e12fd737f91c7e159362488cc2e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198122
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32866}
the limit is ignored anyway. Also rename rtp datachannel
bandwidth limit constant.
BUG=webrtc:6625
Change-Id: If7b26691ced8148955e98c86b9bed692b2e55e8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189972
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32479}
It's currently unused and H265X is not a standardized payload type.
Bug: webrtc:11627
Change-Id: I92e8c7a9eac59ff6d158ed75ae51615c6811cde9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32083}
This reverts commit 20b701f3d7.
Reason for reland: Reverting did not affect the test regression.
Original change's description:
> Revert "sdp: parse and serialize b=TIAS"
>
> This reverts commit c6801d4522.
>
> Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
>
> One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
>
> Original change's description:
> > sdp: parse and serialize b=TIAS
> >
> > BUG=webrtc:5788
> >
> > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31729}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:5788
> Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31762}
TBR=nisse@webrtc.org
Bug: webrtc:5788
Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31816}
These were leftovers from a previous refactoring.
Bug: none
Change-Id: Iee12c2f7f9a7d80ae8e67aa9134ec84894f94960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176327
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31392}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
