Commit Graph

4868 Commits

Author SHA1 Message Date
c7f0dff191 Convert GN libs lists to frameworks
GN recently added support for Apple frameworks to link, rather than
overloading the libs lists. This pulls .frameworks out of the libs
lists, so that GN can stop supporting .frameworks in libs in the
future.

Bug: chromium:1052560
Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31632}
2020-07-06 10:08:09 +00:00
62032d4592 Updates rtp_sender_unitests to use separate thread for pacer calls.
Some classes such as RtpSenderEgress makes assumptions about which
threads (e.g. paced sender vs worker thread) call specific methods.
Unit tests currently are single threaded so these checks are
essentially noops.

This change uses a separate task queue for calls epected to be called
by the pacer, so that inconsistencies in thread can be detected early.

Bug: None
Change-Id: Ic0904304a67eb034033524e62306da34b9eab8b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178602
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31628}
2020-07-03 19:51:58 +00:00
41525d0cc0 red: implement RED with distance 2
Extends the RED implementation to support a distance of two, i.e. two
packets redundancy.

BUG=webrtc:11640

Change-Id: I5113a97a4e3d45d836d7952a0c19c5381069c158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178565
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31625}
2020-07-03 13:53:28 +00:00
1d50cb61d8 Reland "Reland "Allows FEC generation after pacer step.""
This is a reland of 19df870d924662e3b6efb86078d31a8e086b38b5
Patchset 1 is the original.
Subsequent patchset changes threadchecker that crashed with downstream
code.

Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}

Bug: webrtc:11340
Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31619}
2020-07-03 07:20:06 +00:00
1e10a61564 In av1 add structure with custom spatial scalability factor
Bug: webrtc:11042
Change-Id: I93f125fba1fa21d060de47c96435798525f6b374
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178566
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31617}
2020-07-02 14:52:08 +00:00
a1888ae791 Revert "Reland "Allows FEC generation after pacer step.""
This reverts commit 19df870d924662e3b6efb86078d31a8e086b38b5.

Reason for revert: Downstream project failure

Original change's description:
> Reland "Allows FEC generation after pacer step."
> 
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
> 
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
> 
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
> 
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: I3b2b25898ce88b64c2322f68ef83f9f86ac2edb0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178563
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31614}
2020-07-02 12:03:07 +00:00
19df870d92 Reland "Allows FEC generation after pacer step."
This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0

Patchset 2 contains a fix. Old code can in factor call
RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
is not supported there - we shouldn't crash.

Original change's description:
> Allows FEC generation after pacer step.
>
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
>
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
>
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
>
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}

Bug: webrtc:11340
Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31613}
2020-07-02 11:40:55 +00:00
ba0ba71e93 Add 1 sec timer to ModuleRtpRtcpImpl2 instead of frequent polling.
This reduces the number of times we grab a few locks down from
somewhere upwards of around a thousand time a second to a few times.

* Update the RTT value on the worker thread and fire callbacks.
* Trigger NotifyTmmbrUpdated() calls from the worker.
* Update the tests to use a GlobalSimulatedTimeController.

Change-Id: Ib81582494066b9460ae0aa84271f32311f30fbce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177664
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31602}
2020-07-01 11:24:31 +00:00
ed1fb19be2 Reland "Lets PacingController call PacketRouter directly."
This reverts commit 980cadd02c7384397a41c0e334e9f329f3cc5c65.

Reason for revert: Problematic code now fix.

Original change's description:
> Revert "Lets PacingController call PacketRouter directly."
> 
> This reverts commit 848ea9f0d3678118cb8926a2898454e5a4df58ae.
> 
> Reason for revert: Part of changes that may cause deadlock
> 
> Original change's description:
> > Lets PacingController call PacketRouter directly.
> > 
> > Since locking model has been cleaned up, PacingController can now call
> > PacketRouter directly - without having to go via PacedSender or
> > TaskQueuePacedSender.
> > 
> > Bug: webrtc:10809
> > Change-Id: I181f04167d677c35395286f8b246aefb4c3e7ec7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175909
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31342}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10809
> Change-Id: I1d7d5217a03a51555b130ec5c2dd6a992b6e489e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178021
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31563}

TBR=sprang@webrtc.org,srte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10809
Change-Id: I8bea1a5b1b1f618b697e4b09d83c9aac08099593
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178389
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31600}
2020-07-01 09:51:00 +00:00
4b5792cc4a Reland "Reland "Removes lock release in PacedSender callback.""
This is a reland of b46df3da44c42f6e5055c69a8247a344887108ea

Test case for issue that caused revert added:
https://webrtc-review.googlesource.com/c/src/+/178203

Fix for issue that caused revert:
https://webrtc-review.googlesource.com/c/src/+/178207


Original change's description:
> Reland "Removes lock release in PacedSender callback."
>
> This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
>
> Original change's description:
> > Removes lock release in PacedSender callback.
> >
> > The PacedSender currently has logic to temporarily release its internal
> > lock while sending or asking for padding.
> > This creates some tricky situations in the pacing controller where we
> > need to consider if some thread can enter while we the process thread is
> > actually processing, just temporarily busy sending.
> >
> > Since the pacing call stack is no longer cyclic, we can actually remove
> > this lock-release now.
> >
> > Bug: webrtc:10809
> > Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31206}
>
> Bug: webrtc:10809
> Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31332}

Bug: webrtc:10809
Change-Id: I1dba507220316008c0f3b278df4b732011f257eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178384
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31588}
2020-06-30 09:55:00 +00:00
a1163749fd Move BWE_TEST_LOGGING_COMPILE_TIME_ENABLE code to the worker thread.
This is part of moving calls to GetSendRates() to the worker.

Change-Id: Ifb93096a863ddf2669237e7f44af296d0e086b20
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177661
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31582}
2020-06-29 15:51:14 +00:00
096c0b0921 Post stats updates in RtpSenderEgress to the worker.
On the way remove need for lock for
rtp_sequence_number_map_ and timestamp_offset_.

Change-Id: I21a5cbf6208620435a1a16fff68c33c0cb84f51d
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177424
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31581}
2020-06-29 15:35:09 +00:00
e6ac8ff162 Propagate active decode targets bitmask into DependencyDescriptor
Bug: webrtc:10342
Change-Id: I5e8a204881b94fe5786b14e27cefce2fe056e91b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31579}
2020-06-29 12:54:43 +00:00
11b6f6857f Replace slave -> helper, master -> reference
A slight simplification of the NetEq code is also included.

The subtrees below common_audio, modules/audio_coding and
modules/audio_processing were scanned while making this CL.

Bug: webrtc:11680
Change-Id: I33bb1c75b2e3d1c6793fd1c5741ca59f4b6e8455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178361
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31578}
2020-06-29 12:18:05 +00:00
d21f7ab174 Remove media_has_been_sent from RtpState.
The field is unused and the way it's currently laid out in the code,
it maps to a state in the RtpSenderEgress class - which in turn puts
unnecessary threading restrictions on that class.

Bug: webrtc:11581
Change-Id: I41a4740c3277317f33f8e815d8c12c70b355c1db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177426
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31577}
2020-06-29 09:52:44 +00:00
473bbd8131 Remove a timer from ModuleRtpRtcpImpl2 that runs 100 times a second.
The timer fired a Notify call that goes to an object that already
receives callbacks for every packet from RtpSenderEgress.

Further optimizations will be realized by moving ownership
of the stats to the worker thread and then be able to remove
locking in a few classes that currently are tied to those
variables and the callbacks that previously did not come
from the same thread consistently.

We could furthermore get rid of one of these callback interfaces
and just use one.

Bug: webrtc:11581
Change-Id: I56ca5893c0153a87a4cbbe87d7741c39f9e66e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177422
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31575}
2020-06-29 08:09:14 +00:00
d43c3788d7 Delete old TODO item
Bug: webrtc:10198
Change-Id: I47fd6f78b6d5e888eb51e9004d12161cb13d27b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178182
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31572}
2020-06-26 12:11:42 +00:00
54544ec92a In av1 encoder set bitrate per layer when scalability is used.
Bug: webrtc:11404
Change-Id: If779c16ffb55d28d21a0900439cbe2614507e557
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177015
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31570}
2020-06-26 09:26:05 +00:00
8e144aa33f Remove WebRTC usage of //third_party/pymock
Removes usage of Chromium's //third_party/pymock in favor of the version
provided by vpython. This is so that the third_party version can
eventually be removed.

TBR=aleloi@webrtc.org

Bug: chromium:1094489
Change-Id: I68511e11ed1e517c2b6d3bb832090a3c27e480e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#31568}
2020-06-25 18:01:30 +00:00
bf1816170b In Av1 encoder propagate zero bitrate as inactive decode target
Bug: webrtc:10342
Change-Id: I019acb6cca433db1551c22dd139a735ef976cff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178101
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31567}
2020-06-25 16:32:28 +00:00
e3296b6d4c Ignore inactive chains when writing DependencyDescriptor rtp header extension.
To implement rule
"Chains protecting no active decode targets MUST be ignored."
from https://aomediacodec.github.io/av1-rtp-spec/#a44-switching

Bug: webrtc:10342
Change-Id: Ibe5e0b7b6ab8955419d0d9f996c6397f442e1cda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177668
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31566}
2020-06-25 14:59:38 +00:00
118d01ac35 Revert "Reland "Removes lock release in PacedSender callback.""
This reverts commit b46df3da44c42f6e5055c69a8247a344887108ea.

Reason for revert: May cause deadlock.

Original change's description:
> Reland "Removes lock release in PacedSender callback."
> 
> This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
> 
> Original change's description:
> > Removes lock release in PacedSender callback.
> > 
> > The PacedSender currently has logic to temporarily release its internal
> > lock while sending or asking for padding.
> > This creates some tricky situations in the pacing controller where we
> > need to consider if some thread can enter while we the process thread is
> > actually processing, just temporarily busy sending.
> > 
> > Since the pacing call stack is no longer cyclic, we can actually remove
> > this lock-release now.
> > 
> > Bug: webrtc:10809
> > Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31206}
> 
> Bug: webrtc:10809
> Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31332}

TBR=sprang@webrtc.org,srte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10809
Change-Id: I6b06bafad8cd9eeb22107d04b953fd14b8131afa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31564}
2020-06-25 12:41:48 +00:00
980cadd02c Revert "Lets PacingController call PacketRouter directly."
This reverts commit 848ea9f0d3678118cb8926a2898454e5a4df58ae.

Reason for revert: Part of changes that may cause deadlock

Original change's description:
> Lets PacingController call PacketRouter directly.
> 
> Since locking model has been cleaned up, PacingController can now call
> PacketRouter directly - without having to go via PacedSender or
> TaskQueuePacedSender.
> 
> Bug: webrtc:10809
> Change-Id: I181f04167d677c35395286f8b246aefb4c3e7ec7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175909
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31342}

TBR=sprang@webrtc.org,srte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10809
Change-Id: I1d7d5217a03a51555b130ec5c2dd6a992b6e489e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31563}
2020-06-25 09:56:40 +00:00
afdbf8e6f8 H264: Fix stap-a-to-annex-b loop over-read
While converting the aggregated (stap-a) packet transform packet
framing input into an annex-b framing copy, the two loops (both the
required size calculation and the stap-a-to-annex-b copy) may
over-read the input buffer.

In both buffers, `nalu_ptr` follows the input (stap-a) buffer, which
is located in `data`, and whose length is `data_size`. Buffer is read
until `nalu_ptr` reaches the end of the buffer. Issues is that the 5th
line in the loop:

```
    uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
```

This line accesses `nalu_ptr[1]`, which needs to be protected in
the loop condition. Let's assume `data_size = 4`, and that we restart
the loop with `nalu_ptr = data + 3`. The condition of the loop does
hold (`nalu_ptr = data + 3 < data + data_size`), but the 5th line
will access to `data[3+1] = data[4]`, which is an over-read.

Tested:

```
$ ninja -C out/Default
$ out/Default/modules_unittests --gtest_filter=PacketBuffer*:H264*:RtpPacketizerH264Test*:VideoRtpDepacketizerH264Test*:TestH264SpsPpsTracker* --logs
...
[  PASSED  ] 97 tests.
```

Change-Id: I8b8aaf7d12b0bb154430b8922f099cd49e684762
Bug: webrtc:11698
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177140
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31561}
2020-06-24 21:54:45 +00:00
603cc3a31e red: modify the encoder to send RFC 2198
modifies the RED encoder to send the actual RFC 2198 format
described in
  https://tools.ietf.org/html/rfc2198
Decoding is handled in neteq, see red_payload_splitter.h

BUG=webrtc:11640

Change-Id: Ib3005882a3ceee49d2b05c43357f552432a984ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176371
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31560}
2020-06-24 21:25:05 +00:00
1b48532208 Revert "Allows FEC generation after pacer step."
This reverts commit 75fd127640bdf1729af6b4a25875e6d01f1570e0.

Reason for revert: Breaks downstream test

Original change's description:
> Allows FEC generation after pacer step.
> 
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
> 
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
> 
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
> 
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}

TBR=sprang@webrtc.org,srte@webrtc.org

Change-Id: Ie714e5f68580cbd57560e086c9dc7292a052de5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177983
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31559}
2020-06-24 18:41:10 +00:00
75fd127640 Allows FEC generation after pacer step.
Split out from https://webrtc-review.googlesource.com/c/src/+/173708
This CL enables FEC packets to be generated as media packets are sent,
rather than generated, i.e. media packets are inserted into the fec
generator after the pacing stage rather than at packetization time.

This may have some small impact of performance. FEC packets are
typically only generated when a new packet with a marker bit is added,
which means FEC packets protecting a frame will now be sent after all
of the media packets, rather than (potentially) interleaved with them.
Therefore this feature is currently behind a flag so we can examine the
impact. Once we are comfortable with the behavior we'll make it default
and remove the old code.

Note that this change does not include the "protect all header
extensions" part of the original CL - that will be a follow-up.

Bug: webrtc:11340
Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31558}
2020-06-24 16:59:50 +00:00
d95138b684 Make stable target adaptation enabled by default.
This will result in slightly higher encode bitrates and longer frame
lengths compared to using the smoothing filter.

Bug: webrtc:10981
Change-Id: I64704196c56b0ad910895c908baad38c994a971b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177425
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31556}
2020-06-24 11:22:21 +00:00
ee5b6de5aa Add helper for DependencyDescriptor rtp header extension
to decide when to set active_decode_target_bitmask field

Bug: webrtc:10342
Change-Id: I348d7467a72b45651455f4574fe8fda3c77ebbae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177400
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31555}
2020-06-24 10:59:40 +00:00
09f635e587 Delete RtpHeaderExtensionMap::Register deprecated long time ago
Bug: None
Change-Id: I2c3d1243bdfc7c9b6c4cdd326454b5b07ed0dde5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177842
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31552}
2020-06-23 22:55:08 +00:00
24762f207f Fix missing dependencies.
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.

Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
2020-06-23 15:46:34 +00:00
30a3e78794 iSAC encoder: Make it possible to change target bitrate at any time
Not just at construction time.

Bug: webrtc:11704
Change-Id: I952c7dbe20774cc976065c7d2f992a80074ebf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177663
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31550}
2020-06-22 14:59:22 +00:00
09867d37ed Share constants for dependency descriptor rtp header extension
Bug: webrtc:10342
Change-Id: I9c81215569bd1bd96b953faa359f5a3d32c7d0c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177521
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31549}
2020-06-22 11:58:29 +00:00
af0d5bca34 Remove ANA FEC control in Opus encoder.
This has been proven to not be useful.

Bug: chromium:1086942
Change-Id: Ib71b194f59301851791a1a056f5f10b98c5a1d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177520
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31548}
2020-06-22 11:18:26 +00:00
7a82467d0d Fix vp9 svc singlecast mode and enable quality scaler for vp9
1) Fix several typos and small mistakes which could lead to crashes
2) Adjust bitrates if leading layers are disabled
3) Wire up webrtc quality scaler

Bug: webrtc:11319
Change-Id: I16e52bdb1c315d64906288e4f2be55fe698d5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177525
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31546}
2020-06-18 21:24:46 +00:00
79ca92d952 Add workaround method for deprecated code.
This is to allow downstream cases to be able to set the
media_has_been_sent flag in the sender as it's being
removed from RtpState.

Bug: webrtc:11581
Change-Id: I28f5fca96ba1d3f562c4d069d1b6d9af4002aaab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177524
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31545}
2020-06-18 17:08:44 +00:00
f46902c540 Add a simple frame length controller.
This will be used when adaptivePtime is enabled.

Bug: chromium:1086942
Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31544}
2020-06-18 15:21:48 +00:00
c5c878b8ab vp9_impl: Enable VP9D_SET_LOOP_FILTER_OPT for libvpx vp9 decoder
enable this opt can give 20% performance improvement for video
decoding with 720P video loopback and fake camera on chromebook sarien.

Bug: None
Test: ./modules_tests on chromebook sarien
Change-Id: I8c6487b291b5861e6ba6b6d55b24d7ddb51c341e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177335
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31543}
2020-06-18 08:32:40 +00:00
969ccf0e12 Reland "VP9 decoder: Sets thread count based on resolution, reinit on change."
This is a reland of d5925756980f6e82a55f57532c8d855e954459fb

Patchset 2 is a reland of
https://webrtc-review.googlesource.com/c/src/+/177012

Patchset 3 is a fix for a potential crash when InitDecode()is called from
VideoStreamDecoderImpl::GetDecoder(), where the decoder_settings
parameter is a but surprisingly set to nullptr.

Original change's description:
> VP9 decoder: Sets thread count based on resolution, reinit on change.
>
> Previously, number of decoder threads for VP9 were always set to 8 but
> with a cap at number of cores. This was done since we "can't know" the
> resolution that will be used.
>
> With this change, we now intialize the number of threads based on
> resolution given in InitDecode(). If a resolution change happens in
> flight, it requires a keyframe. We therefore parse the header from
> any key frame and if it has a new resolution, we re-initialize the
> decoder.
>
> The number of threads used is based on pixel count. We set one thread
> as target for 1280x720, and scale up lineraly from there. The 8-thread
> cap is gone, but still limit it core count.
>
> This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.
>
> Bug: webrtc:11551
> Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31507}

Bug: webrtc:11551
Change-Id: I2b4b146d0b8319f07ce1660202d6aa4b374eb015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177246
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31527}
2020-06-15 19:14:52 +00:00
ee4f5cca00 Wires up trial for smarter postproc settings in non-arm VP8 decoder.
Params and format is the same as for existing ARM experiment, but a new
group name is created for non-ARM experiment.

Bug: webrtc:11551
Change-Id: I3a6c0f07a8c1d714477ae4703c16e48df36ac10e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177102
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31524}
2020-06-15 15:55:39 +00:00
90dc5dae2c Factor encoded frame generation code into own target
Show it can make vp9 tests cleaner too.

Bug: None
Change-Id: I8333a61dec1ef90ade9faffea94e1555ccbfcfaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177013
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31523}
2020-06-15 12:25:06 +00:00
9465978a3b Remove framemarking RTP extension.
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +00:00
ef377ec6d5 Revert "VP9 decoder: Sets thread count based on resolution, reinit on change."
This reverts commit d5925756980f6e82a55f57532c8d855e954459fb.

Reason for revert: May cause crashes.

Original change's description:
> VP9 decoder: Sets thread count based on resolution, reinit on change.
> 
> Previously, number of decoder threads for VP9 were always set to 8 but
> with a cap at number of cores. This was done since we "can't know" the
> resolution that will be used.
> 
> With this change, we now intialize the number of threads based on
> resolution given in InitDecode(). If a resolution change happens in
> flight, it requires a keyframe. We therefore parse the header from
> any key frame and if it has a new resolution, we re-initialize the
> decoder.
> 
> The number of threads used is based on pixel count. We set one thread
> as target for 1280x720, and scale up lineraly from there. The 8-thread
> cap is gone, but still limit it core count.
> 
> This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.
> 
> Bug: webrtc:11551
> Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31507}

TBR=ilnik@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11551
Change-Id: Id235c8ded83b3e1fc1d132c8f56c9f20001f6f22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177242
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31521}
2020-06-15 10:11:09 +00:00
aa5c1e8fd2 Revert "Adjusts allowable thread count for vp9 decoders."
This reverts commit 26e50469513a3df2abb88c9ae2a5af70de0b029c.

Reason for revert: Part of change that may cause crashes.

Original change's description:
> Adjusts allowable thread count for vp9 decoders.
> 
> Set 2 thread as target for 1280x720 pixel count, and then scale up
> linearly from there - but cap at physical core count.
> For common resolutions this results in:
> 1 for 360p
> 2 for 720p
> 4 for 1080p
> 8 for 1440p
> 18 for 4K
> 
> Bug: webrtc:11551
> Change-Id: I666bd971eccddee096749f20d3b08eb40fe868ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177012
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31513}

TBR=sprang@webrtc.org,kron@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11551
Change-Id: I4ea5166efeed3d55255a0243a69deb584a0e19e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31517}
2020-06-15 08:38:54 +00:00
55afe3885b Search and replace gendered terms according to style guide:
https://chromium.googlesource.com/chromium/src/+/master/styleguide/inclusive_code.md#tools

Not changin the transcipt in
resources/audio_processing/conversational_speech/README.md

BUG=webrtc:11680

Change-Id: I36af34e4a4e0ec6161093c0045b7bbe1dbe4eb45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177016
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31514}
2020-06-12 14:12:54 +00:00
26e5046951 Adjusts allowable thread count for vp9 decoders.
Set 2 thread as target for 1280x720 pixel count, and then scale up
linearly from there - but cap at physical core count.
For common resolutions this results in:
1 for 360p
2 for 720p
4 for 1080p
8 for 1440p
18 for 4K

Bug: webrtc:11551
Change-Id: I666bd971eccddee096749f20d3b08eb40fe868ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177012
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31513}
2020-06-12 13:52:19 +00:00
222fdfdace in AV1 Encoder remove hack of removing temporal delimiter OBU
Bug: webrtc:11174
Change-Id: I70e3ab1af92562b44f974aa415e0215e06fcf1cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177009
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31512}
2020-06-12 12:42:09 +00:00
7ff6355b88 Add decoder support for VP9 profile 1 I444
libvpx already supports VP9 profile 1. Add code to enable SDP negotiation of receiving VP9 profile 1.

Bug: webrtc:11555
Change-Id: I35d12d159a1414aac744f202331d3a9c4a84f5af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176322
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31509}
2020-06-12 05:17:24 +00:00
b64ecd9960 Check V4L2_CAP_VIDEO_CAPTURE when enumerating capture devices on Linux
The side effect of not filtering on V4L2_CAP_VIDEO_CAPTURE is that
every device is enumerated twice. Because we look up devices by name,
and the device that supports V4L2_CAP_VIDEO_CAPTURE seems to always
appear first in /dev/video, this does not seem to end up with us ever
choosing an inappropriate device. We might get away with just filtering
device names from the list, but if the order of devices ever changed in
/dev/video there could be problems.

Bug: webrtc:11641
Change-Id: I16fee4edc873838ed4643ee16a8bbc699d6bbcf5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176460
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Dan Minor <dminor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31508}
2020-06-11 19:58:49 +00:00
d592575698 VP9 decoder: Sets thread count based on resolution, reinit on change.
Previously, number of decoder threads for VP9 were always set to 8 but
with a cap at number of cores. This was done since we "can't know" the
resolution that will be used.

With this change, we now intialize the number of threads based on
resolution given in InitDecode(). If a resolution change happens in
flight, it requires a keyframe. We therefore parse the header from
any key frame and if it has a new resolution, we re-initialize the
decoder.

The number of threads used is based on pixel count. We set one thread
as target for 1280x720, and scale up lineraly from there. The 8-thread
cap is gone, but still limit it core count.

This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.

Bug: webrtc:11551
Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31507}
2020-06-11 17:43:27 +00:00