When enabled:
- Creates an audio network adapter config that is passed to audio send
stream.
- Configures a lower default min bitrate.
All parameters can be configured via a field trial that can also force
enable the audio network adaptor (this is mainly intended for testing).
Bug: chromium:1086942
Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31565}
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}
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}
negotiates the RED codec for opus audio behind a field trial
WebRTC-Audio-Redundancy
This adds the following line to the SDP:
a=rtpmap:someid RED/48000/2
To test start Chrome with
--force-fieldtrials=WebRTC-Audio-Red-For-Opus/Enabled
BUG=webrtc:11640
Change-Id: I8fa9fb07d03db5f90cdb08765baaa03d3d0458cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176372
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31562}
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}
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}
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}
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}
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}
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}
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}
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}
WebRTC doesn't use these features, so disable them to reduce the
potential attack surface.
Bug: webrtc:11694
Change-Id: I093aa824c6da592852270534ae7415ceb19fca47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31540}
This is needed because chromium build targets need to be exported for
its component builds.
// TBR because this is a purely building related change and it has been
// reviewed by mbonadei@.
TBR=stefan@webrtc.org
Bug: webrtc:11525
Change-Id: I97f0c814b11e7fad86eeff319e644ae51204c3b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31534}
This removes code from DataChannelController that exposes
an internal vector of data channels and puts the onus of
returning stats for a data channel, on the data channel
object itself. This will come in handy as we make threading
changes to the data channel object.
Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639
Bug: webrtc:11547, webrtc:11687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31533}
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}
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}
