This reverts commit 75990b9a8f98ea2d597a31472fb778ec4d55f698.
Reason for revert: Breaks downstream, a use case of having three VP9
encodings, scalability mode only specified on the first layer
(L2T2_KEY) and the other two layers not having a scalability mode but
also being active=false appears to trigger a DCHECK in
call/rtp_video_sender.cc:501. More investigation needed
Original change's description:
> Ship ability to opt-in to VP9/AV1 simulcast.
>
> With this unflagging, an app can opt-in to simulcast when using multiple
> encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
> ensures backwards-compat with apps relying on 3 encodings to mean SVC
> who traditionally have not specified scalabilityMode.
>
> It fixes the spec/API bug of asking for simulcast and not getting
> simulcast. The field trial exists only as a kill-switch with a TODO to
> remove it.
>
> This ships initial support, however note that the VP9/AV1 simulcast uses
> SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
> allocator uses more kbps than SvcRateAllocator. This should be revisited
> to avoid significant higher bitrates, for example when comparing VP9
> simulcast to VP9 SVC.
>
> Shipping the ability for apps to opt-in makes it easier to exercise
> these new code paths and allows initial feedback from developers, but
> due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
> many apps may find that VP9 SVC is still more beneficial for BW reasons.
>
> Bug: webrtc:14884, webrtc:15005
> Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39601}
Bug: webrtc:14884, webrtc:15005
Change-Id: Ic8f77e6a2971f493d6cd8c23faecd435058a8847
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298440
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39605}
With this unflagging, an app can opt-in to simulcast when using multiple
encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
ensures backwards-compat with apps relying on 3 encodings to mean SVC
who traditionally have not specified scalabilityMode.
It fixes the spec/API bug of asking for simulcast and not getting
simulcast. The field trial exists only as a kill-switch with a TODO to
remove it.
This ships initial support, however note that the VP9/AV1 simulcast uses
SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
allocator uses more kbps than SvcRateAllocator. This should be revisited
to avoid significant higher bitrates, for example when comparing VP9
simulcast to VP9 SVC.
Shipping the ability for apps to opt-in makes it easier to exercise
these new code paths and allows initial feedback from developers, but
due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
many apps may find that VP9 SVC is still more beneficial for BW reasons.
Bug: webrtc:14884, webrtc:15005
Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39601}
Because the adapter has a passthrough mode, it can already handle both
singlecast and simulcast cases, meaning the proxy is no longer providing
value. Let's delete.
Bug: webrtc:15001
Change-Id: I480eaba599448e9b82b8cf7f829dc35ad6ce0434
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39579}
The goal of the VP9 simulcast project is that when `scalability_mode`
is set, multiple encodings are always interpreted as simulcast, even
if VP9 or AV1 is used. This CL makes this so, but only if the flag
"WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us
to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast.
When we are ready to ship we will remove the need to use the field
trial, but before we ship this we'll want to revisit if
SvcRateAllocator can be updated to support simulcast. (Today if we use
SvcRateAllocator when VP9 simulcast is used, all encodings except the
first one get bitrate=0, causing the test to fail because media is not
flowing on all layers.) For now, a TODO is added.
Bug: webrtc:14884
Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39552}
Encoders that do not support simulcast in the first place does not
expect to have to handle simulcast configurations, and as such may not
necessarily return
WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED from
InitEncode().
This CL updates EncoderSimulcastProxy to respect this info to avoid
silent errors when LibvpxVp9Encoder (which does not support simulcast)
is attempted to be used in simulcast.
Alternatively we can try to get rid of EncoderSimulcastProxy altogether
since SimulcastEncoderAdapter already has a passthrough mode. A TODO is
added to get rid of the proxy.
Bug: webrtc:14884
Change-Id: Id3703f1768b0aebf617b7d9b935914cd5f1b0f52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296885
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39541}
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808.
Reason for revert: Added a field trial that allows landing the code without affecting performance in prod.
This CL also incorporates subsequent CLs that also had to be reverted.
Original change's description:
> Revert "Use two MediaChannels for 2 directions."
>
> This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
>
> Reason for revert: Quality regression detected.
>
> Original change's description:
> > Use two MediaChannels for 2 directions.
> >
> > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
> >
> > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
> >
> > Bug: webrtc:13931
> > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39340}
>
> No-Try: true
> Bug: webrtc:13931
> Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39445}
Bug: webrtc:13931
Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39494}
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread. Avoid this
by posting only if invoked on a diffferent sequence.
TESTED=presubmit + local Meet calls.
Bug: webrtc:137439
Change-Id: Ic5dd99e5fbb843ad4c54d4466138135ae81596cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295867
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39471}
This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
Reason for revert: Quality regression detected.
Original change's description:
> Use two MediaChannels for 2 directions.
>
> This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
>
> The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
>
> Bug: webrtc:13931
> Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39340}
No-Try: true
Bug: webrtc:13931
Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39445}
This reverts commit a087f6f1c842f1d70ad207b44c48321ab60d2d95.
Reason for revert: Needed to roll back other CL
Original change's description:
> Add plumbing for video NACK to be coupled between channels.
>
> Bug: webrtc:13931, webrtc:14920
> Change-Id: I451869e295e099a1d08c0c80e481decd53149f1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294382
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39373}
Bug: webrtc:13931, webrtc:14920
Change-Id: I19e176e75630313da470542e7ff1e89b6d717fc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295664
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39432}
Currently, send stats update `last_stats_log_ms_` causing receive stats
to never be logged.
This behavior was introduced in https://webrtc-review.googlesource.com/c/src/+/288750
Bug: b/270519075
Change-Id: Ie781082cfb212c1c903cbada5e393d2e7aa6150f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294743
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39381}
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}
This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
Bug: webrtc:13931
Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39340}
As part of go/unblocking-vp9-simulcast (Step 1), EncodedImage is being
upgraded to be able to differentiate between what is a simulcast index
and what is a spatial index.
In order not to break existing code assuming that "if codec != VP9,
SpatialIndex() is the simulcast index", SimulcastIndex() has fallback
logic to return the value of spatial_index_ in the event that
SetSimulcastIndex() has not been called. This allows migrating external
code from (Set)SpatialIndex() to (Set)SimulcastIndex(). During this
intermediate time, codec gates are still necessary in some places of
the code, see TODOs added.
In a follow-up CL, after having fixed dependencies, we'll be able to
remove the fallback logic and rely on SimulcastIndex() and
SpatialIndex() actually being the advertised index and "if codec..."
hacks will be a thing of the past!
Bug: webrtc:14884
Change-Id: I70095c091d0ce2336640451150888a3c3841df80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293343
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39318}
This allows MediaChannel to know whether it's being used
for sending, receiving, or both. This is a preparatory CL
for landing the split of MediaChannel usage into sending and
receiving objects.
Bug: webrtc:13931
Change-Id: If518c8b53d5256771200a42e1b5f2b3321d26d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292860
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39283}
This is used when an unsignaled stream with a known payload type is received and later a RTX packet is received.
Bug: webrtc:14817
Change-Id: I29f43281cec17553e1ec2483e21b8847714d2931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291328
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39243}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
This ensure transport feedback is sent for RTX packets that are received before media payload packets.
Bug: webrtc:14795, webrtc:14817
Change-Id: I6a2579b87c8863e003decb2b2559ef51a852cadb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291119
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39174}
also refactor the code to have FillSendCodecStats/FillReceiveCodecStats
methods for similarity with the video engine code.
BUG=webrtc:14808
Change-Id: Ib0687f36a4b4a71c849e0b4918e50592d7772ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290891
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39172}
optimizing for the fairly common case of many recv-only
mediasections.
BUG=webrtc:14808
Change-Id: Iae68c5bb7a5516d77f908f1effbb50a5ed750f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290984
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39122}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
instead of the full set of codecs that have been negotiated.
BUG=webrtc:14808
Change-Id: I464cc1d20e5b5227a09929c909615b432c6be041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39114}
Fallback to a default value if the scalability mode is unset or not supported by the codec.
The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set).
Bug: webrtc:11607
Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39074}
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.
Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
This is done in peparation for using these methods in the engines.
Bug: webrtc:7135, webrtc:14795
Change-Id: I1255c035437d31398327318c3dbd73e70a11a5cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290577
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39044}
This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523
See diff between Patch Set 1 and latest Patch Set.
The original CL broke this WPT[1] because getStats() with the receiver
as the selector stopped working in the event of unsignalled SSRCs due
to the receiver not knowing what the SSRC was.
This fix is to query media_channel_ for the unsignalled SSRC in the
event that the receiver does not know the SSRC.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175, webrtc:14811
Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39010}
This is in preparation for splitting MediaChannel into sender and
receiver channels, with independent objects.
Bug: webrtc:13931
Change-Id: I8e34b0c80b4d76132394efcda658a8face3ab873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288750
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38998}
This is a reland of commit 97ba853295578975a04fc504315cccd465f9f0bd
This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502.
Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}
Bug: webrtc:14802
Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38997}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
This CL removes a couple more opportunities for client code
to interact directly with the MediaChannel implementation classes.
No-try because of infra failure.
No-Try: true
Bug: webrtc:13931
Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38935}
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.
Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
Passes frame_types to the underlying encoder in bypass mode.
For libvpx this has no effect, for H264 this changes the behavior
to allow generating keyframes on a per-layer basis.
BUG=chromium:1354101
Change-Id: I26fc22d9e2ec4681a57ce591e9eafd0b1ec962b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285083
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38821}
