Commit Graph

4291 Commits

Author SHA1 Message Date
97ffbefdab Pass and store PacketBuffer::Packet by unique_ptr
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.

Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
2020-01-29 11:48:55 +00:00
48be482d73 Fix spelling.
Bug: None
Change-Id: Id281fe3d58bd5a8651b299b426353524085dd876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167536
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rikard Lundmark <lundmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30402}
2020-01-29 10:50:14 +00:00
71a77c4b3b Adds trial to use correct overhead calculation in pacer.
Bug: webrtc:9883
Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30399}
2020-01-29 09:39:40 +00:00
b6bf0b2546 Pass picture_id from generic packetizer through codec-specific field
To free up RtpVideoHeader::generic field for codec agnostic details
from an rtp header extension.

Bug: webrtc:10342
Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30396}
2020-01-28 19:26:28 +00:00
260c788d77 AEC3: Added multi-channel support for the capture delay functionality
This CL adds the missing support for multi-channel in the code that
provides an optional and configurable delay to be added to the
microphone signal.

The CL also makes the creation of the delay object conditional on the
need for that support (this is important since this adds a significant
heap memory footprint)

Bug: webrtc:11314,chromium:1045910
Change-Id: I92d577e31af830945fe9d5ca2032000aad4266be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167525
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30392}
2020-01-28 15:39:26 +00:00
086055d0fd Reland "Only include overhead if using send side bandwidth estimation."
This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
2020-01-28 10:36:39 +00:00
c709412c76 Revert "Only include overhead if using send side bandwidth estimation."
This reverts commit 8c79c6e1af354c526497082c79ccbe12af03a33e.

Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.

Original change's description:
> Only include overhead if using send side bandwidth estimation.
> 
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}

TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org

Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
2020-01-27 15:09:49 +00:00
8c79c6e1af Only include overhead if using send side bandwidth estimation.
Bug: webrtc:11298
Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30382}
2020-01-27 14:19:54 +00:00
ff0e4dbd1f Reland "Send absolute capture time through audio coding module."
This is a reland of 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a

Original change's description:
> Send absolute capture time through audio coding module.
>
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

Bug: webrtc:10739
Change-Id: I10086d239ad3f1efb8485098bf3b0ad23110962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167213
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30380}
2020-01-27 13:18:27 +00:00
6c9bc396e9 Cleanup log formatting in modules/audio_processing
Bug: None
Change-Id: I47177530d8a85d7b2f143081de71f5a3bf8ec354
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166041
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30379}
2020-01-27 09:42:56 +00:00
f3886aea86 Include cursor rects in updated_region.
DesktopAndCursorComposer adds the cursor image to the desktop, but does
not change the updated_region, so it generally doesn't encode correctly
unless the mouse is moving over a region that is changing. This CL
extends the updated region to include the union of the old and new
cursor rects, with an optimization for the case where the cursor has
neither moved nor changed.

Bug: chromium:1043325
Change-Id: I52076c96528820833fda6aa95f5b1fbc0f613909
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166545
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30374}
2020-01-24 20:16:58 +00:00
b039c30157 Reland "Change log level of AEC3 buffer info to VERBOSE"
This is a reland of 48148dc840f66c5f6adc5e2ba01c15104e0a9bab

Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
>
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
>
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}

Bug: webrtc:11278, webrtc:11295
Change-Id: I8e6f11457e283c83cae5581adcacdc4d3b5431bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30372}
2020-01-24 12:58:08 +00:00
b18c4eb0a9 Add parameterization for three multi channel AEC3 unit tests
Bug: webrtc:11295
Change-Id: I478aa02908c494cf9609db00021438a59a132b66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167202
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30370}
2020-01-24 12:26:46 +00:00
159c414ff8 Detach LossNotificationController from RtpGenericFrameDescriptor
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
2020-01-24 11:53:28 +00:00
88636c6dac Improvements for NetEqControllers
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.

Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
2020-01-24 11:39:52 +00:00
760fd52494 Replace MockAudioDeviceModule mock refcounting with real refcounting
Bug: webrtc:11308
Change-Id: Ic55ec2c4b45f8fc709fe1348556bdeea6202e7a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166580
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30366}
2020-01-23 19:04:58 +00:00
4175914f41 Revert "Send absolute capture time through audio coding module."
This reverts commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a.

Reason for revert: failing upstream tests

Original change's description:
> Send absolute capture time through audio coding module.
> 
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com

Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
2020-01-23 16:21:06 +00:00
48655cfdbf Send absolute capture time through audio coding module.
Bug: webrtc:10739
Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30363}
2020-01-23 16:06:12 +00:00
3c7e4dd85f Revert "Change log level of AEC3 buffer info to VERBOSE"
This reverts commit 48148dc840f66c5f6adc5e2ba01c15104e0a9bab.

Reason for revert: Causing tests to timeout, see bugs.webrtc.org/11295

Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
> 
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
> 
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}

TBR=saza@webrtc.org,peah@webrtc.org

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

Bug: webrtc:11278
Change-Id: I283648a6d4d58cfe7af7a646d915122207883007
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30357}
2020-01-23 10:28:25 +00:00
5bb9adcb08 Add absolute capture time to video sender path.
Bug: webrtc:10739
Change-Id: I2bbef7275ae065312ad86daaecc773c0ab36a684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167061
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30344}
2020-01-22 13:09:28 +00:00
39c8350613 Reduce the complexity of the multichannel echo subtractor test
This CL reduces the complexity of the Subtractor.ConvergenceMultiChannel
test by
1. Slightly reducing the amount of tested combinations for the non-debug
   mode.
2. Drastically reduce the amount of tested combinations for the debug
   mode.


Bug: webrtc:11295
Change-Id: I56bfa4a1463d26e5217b6a4d7f2ef54de7aab512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166529
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30343}
2020-01-22 11:39:07 +00:00
d74c56fcd0 Add absolute capture time to audio sender path.
WebRTC prototype:
https://webrtc-review.googlesource.com/c/src/+/158520

Bug: webrtc:10739
Change-Id: I07b7a60602b41dc04292a91923e878a8d753486f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161732
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30335}
2020-01-21 13:06:18 +00:00
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00
4bab2fcf6b [Overuse] Setting encoder configurations through the interface.
This squashes together several input signals that were spread out
through several calls into a single method and calling place:
SetEncoderSettings(), invoked from ReconfigureEncoder(). This is added
to the abstract interface.

This makes the following methods obsolete which are removed:
- SetEncoder(): The VideoEncoder was only used for GetEncoderInfo();
  the VideoEncoder::EncoderInfo is now part of the EncoderSettings.
- SetEncoderConfig(): The VideoEncoderConfig is part of
  EncoderSettings. The config is used for its codec_type and
  content_type enums.
- SetCodecMaxFrameRate(): The max frame rate was the same as
  VideoCodec::maxFramerate. VideoCodec is now part of EncoderSettings.

There may be some overlap in information between EncoderConfig and
VideoCodec, but that is outside the scope of this CL, which only makes
sure to bundle encoder settings-like information into one input signal.

Bug: webrtc:11222
Change-Id: I67c49c49c0a859cb7d5051939a461593c695a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166602
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30332}
2020-01-21 11:48:11 +00:00
67dcb4b54d Publish DependencyDescriptor structures in the api
The extension (and thus structures to carry it) are designed
in particular for client<->SFU link. Putting the structure into api
acknowledges it can be reused by SFU projects

Bug: webrtc:10342
Change-Id: I8ca1f5046abadf6aa16200443c4892e9a2a928b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166467
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30324}
2020-01-20 15:05:48 +00:00
cea929923b in RtpPacket packet pass rtp header extension value by const&
to allow writing DependencyDescriptor value that is not copiable.
and avoid copying RtpGenericFrameDescriptor

Bug: webrtc:10342
Change-Id: I6eefa9d06b90d7e858f224443ba6769975b556fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166171
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30322}
2020-01-20 13:37:01 +00:00
cee751abff Reland "Enable using a custom NetEqFactory in simulations"
This is a reland of 2a11b2451a4068746fa0c55fa210efd4a15e4423
There are no changes compared to the first attempt.

Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}

TBR=kwiberg

Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
2020-01-20 12:46:34 +00:00
629de6f7ed Merge RtpPacket HasExtension and IsExtensionReserved functions
RtpPacket doesn't keep difference between reserved and set extension.

Bug: None
Change-Id: I1c79f4ebd7ba20ae5da0194c3faa418050db7d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30316}
2020-01-20 11:37:25 +00:00
c380e97ee6 Make MouseCursorMonitor optional for DesktopAndCursorComposer.
DesktopAndCursorComposer already handles a null MouseCursorMonitor. This
CL allows that code-path to be utilized by callers that already have a
MouseCursorMonitor, allowing its callbacks to be re-used by this class.
This is more efficient, and works around an apparent X Server deadlock
on Linux if multiple MouseCursorMonitors are simultaneously active.

The intended use-case for this is to allow the host-side cursor to be
composited into the desktop image if mouse-lock is active at the client.

Bug: chromium:1043325
Change-Id: I7e036850dd8c17fe55e57db252392062a847d10f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166581
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30312}
2020-01-18 01:28:22 +00:00
4b47dd39a7 Make deprecated OnMouseCursorPosition overload optional.
The only callers or non-trivial implementations of this that I could
find are in remoting/ in Chromium, which I plan on fixing once this
gets rolled.

Bug: chromium:1043325
Change-Id: Id5a33fc09bb066f979876b2a7dcbc3dc5c2d3dd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166560
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30310}
2020-01-17 23:14:21 +00:00
7a709c0e85 RtpReferenceFrameFinder: protect against crashes due to large temporal idx value on the wire
Bug: chromium:1042933
Change-Id: Ide37812a73b72e744f45b671918dc9817775e1f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166463
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30307}
2020-01-17 16:24:53 +00:00
df2c601616 Move Offset constants from VideoSendTiming value to VideoTimingExtension class
These constants describes how value should be put on the wire and thus
belong to the extension builder/writer class rather than extension value class

Bug: None
Change-Id: I65ca3923eddcc2e48563ad69b98356c159ad86be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30305}
2020-01-17 15:57:38 +00:00
db6ca7f2d7 Add safety checks in RtpPacket::ZeroMutableExtensions and fuzz it
Bug: chromium:1042535
Change-Id: I0f7ef1086631b5beb2e0c89d57534d2551289117
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166441
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30303}
2020-01-17 14:22:04 +00:00
c9f42ad909 Simplifies transport overhead mechanism in Scenario test framework.
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.

Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
2020-01-17 11:30:02 +00:00
2d31aea481 Remove unused AEC delay offset API
Bug: webrtc:5298
Change-Id: If490dba3c95b1d6aeaa7b110dd1ffc23ee7a96c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166440
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30297}
2020-01-17 11:09:51 +00:00
0695df1a59 Reland "Replace the ExperimentalAgc config with the new config format"
This is a reland of f3aa6326b8e21f627b9fba72040122723251999b

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
2020-01-17 10:09:09 +00:00
3f0bc2c176 Revert "Enable using a custom NetEqFactory in simulations"
This reverts commit 2a11b2451a4068746fa0c55fa210efd4a15e4423.

Reason for revert: Causes b/147826709

Original change's description:
> Enable using a custom NetEqFactory in simulations
> 
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}

TBR=kwiberg@webrtc.org,ivoc@webrtc.org

Change-Id: I14a0bd6ad2a90f1686b8b1a78f18aea9325871fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11005
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166403
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Sandeep Siddhartha <sansid@google.com>
Cr-Commit-Position: refs/heads/master@{#30288}
2020-01-16 22:56:21 +00:00
2a11b2451a Enable using a custom NetEqFactory in simulations
Bug: webrtc:11005
Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30286}
2020-01-16 18:26:44 +00:00
658f1814da Reland "Moves TransportFeedbackAdapter to TaskQueue."
This is a reland of 62d01cde6f6ec1fa91b1e5234a7922ad1a4ce036

Original change's description:
> Moves TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30037}

Bug: webrtc:9883
Change-Id: Icc63883903b283d490e9d4ed455e0eca69ed2074
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162000
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30285}
2020-01-16 16:41:53 +00:00
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
05f8487627 Add processing time to VideoFrame
Bug: chromium:1011581
Change-Id: Icd675cb98b8b5052933b9a8eebe718be94c2fef2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166162
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30281}
2020-01-16 14:11:15 +00:00
c6f81a71e5 Remove higher_spatial_layers from RTPVideoHeader structure as unused.
The idea to communicate spatial dependencies with spatial layers bitmask
wasn't fully implemented and was dropped in later version of the descriptor.

Bug: webrtc:10342
Change-Id: I1ed191c3a2a9d2e1e9ddf313f781ca8257c34dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166165
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30278}
2020-01-16 11:11:39 +00:00
6ca908f48c Shorten the fir filter adapt test quite a bit.
The test is likely timing out on iOS simulator (see bug). Maybe I'm
going a bit overboard here :) if you want to keep all the cases I
removed, you can run some cases in one test method and the others in
another test method. Are the cases I removed particularly important?

Bug: webrtc:11284
Change-Id: I8f2e8830f931594c3471d1c20a2654e258b9fcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166169
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30277}
2020-01-16 09:12:26 +00:00
edb80cff01 Delete RtpDepacketizer interface as no longer used
Bug: webrtc:11152
Change-Id: I0c5f2167ba39c22f4491d2e34f3462b9ecb9bf2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166160
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30276}
2020-01-16 09:00:16 +00:00
ea992f8771 Reland "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
This reverts commit d2d7a47247187236ce62e3c842963f6e4e9f0f1f.

Reason for revert: This revert is not needed. Failure was not due to webrtc.

Original change's description:
> Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
> 
> This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.
> 
> Reason for revert: Causing a build break:
> webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
> this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
>   'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'
> 
> 
> 
> Original change's description:
> > Reland "Extracts ssrc based feedback tracking from feedback adapter."
> > 
> > This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> > 
> > Original change's description:
> > > Extracts ssrc based feedback tracking from feedback adapter.
> > > 
> > > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > > 
> > > Bug: webrtc:9883
> > > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30076}
> > 
> > Bug: webrtc:9883
> > Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30266}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9883
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30270}

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

Change-Id: Idd1073ebfef77b2154d7123b47dacb479537c550
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166200
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30271}
2020-01-15 18:24:32 +00:00
d2d7a47247 Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.

Reason for revert: Causing a build break:
webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
  'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'



Original change's description:
> Reland "Extracts ssrc based feedback tracking from feedback adapter."
> 
> This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> 
> Original change's description:
> > Extracts ssrc based feedback tracking from feedback adapter.
> > 
> > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > 
> > Bug: webrtc:9883
> > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30076}
> 
> Bug: webrtc:9883
> Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30266}

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

Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30270}
2020-01-15 17:44:42 +00:00
d61338fa6e Reland "Extracts ssrc based feedback tracking from feedback adapter."
This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974

Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
> 
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}

Bug: webrtc:9883
Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30266}
2020-01-15 12:51:16 +00:00
61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
6ef59d1ced Don't pace audio by default
After experimentation, not pacing audio is better. This is controlled
by the field trial WebRTC-Pacer-BlockAudio. This change keeps the flag,
but changes the behaviour such that it defaults to Disabled. However,
audio can still be paced if one chooses by enabling the field trial.

Bug: webrtc:11257
Change-Id: I5b23a82bb6708c007cf8dfb40065c821eefdc4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165381
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30262}
2020-01-15 11:21:14 +00:00
d06588a758 Change Av1 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00