Commit Graph

4005 Commits

Author SHA1 Message Date
6a05bb1b12 AEC3: Add signal dependent mixing before alignment
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.

As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.

Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
2019-12-03 11:30:48 +00:00
3a77f93589 AEC3: Ensure that the high-pass filter effect is on when AEC3 is active
This CL ensures that the high-pass filter is on whenever the echo
controller is on. This is important as the echo controller code assumes
that the external high-pass filter is active.

The CL also corrects the ToggleAec unit test (which started failing
after this code change).

Bug: webrtc:11159,chromium:1030179
Change-Id: Ie29db74bf3de6279a08564398d32d67d5e1569db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161222
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29979}
2019-12-03 11:06:08 +00:00
21021f022b NetEq: Fix bug in PLC for multi-channel audio
There is currently a bug in NetEq that causes audio to leak from the
first channel to all others during loss concealment. This CL fixes the
problem and also adds a unit test to verify.

Bug: webrtc:11145
Change-Id: Ia6c4a234ff7f78e9a6080f1cb17eb80af671c3dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161091
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29974}
2019-12-02 17:44:58 +00:00
5256d8bc4b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect
Bug: webrtc:10138
Change-Id: I22079e2630bb1f3bb27472795fe923f9143b3401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161010
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29973}
2019-12-02 17:11:37 +00:00
b2b58d84e3 AEC3: Adding default AEC3 configurations that are setup specific
This adds functionality to AEC3 to produce setup-specific
default configurations that are tailored to work well for the
number of channels at hand.

The tunings are only used for the case when no echo control factory
has been provided.

Bug: webrtc:11151,chromium:1029717
Change-Id: I1bd2d10327300c7b0f3169a52bf66700b781fd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161086
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29972}
2019-12-02 15:04:07 +00:00
cf20519262 AEC3: Correct the number of render channels in the echo audibility code
Bug: webrtc:11150,chromium:1029707
Change-Id: I4d43bfcd52871a45e7608158bf19c32523976f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29971}
2019-12-02 13:13:11 +00:00
b144c58973 Remove deprecated setting for activating multichannel processing
Bug: webrtc:10859
Change-Id: I86cae6a9b765bc807c00632ec7d743b754941f81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160780
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29966}
2019-12-02 07:47:53 +00:00
39cf3c723e Clean up the NetEqFactory API.
This CL decouples NetEqFactory and AudioDecoderFactory.
AudioDecoderFactory is used in more places than just inside of NetEq, so
decoupling these makes sense.

Bug: webrtc:11005
Change-Id: I78dd856e4248e398e69a65816b062ef30555b055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29961}
2019-11-29 14:04:44 +00:00
2d02c943b2 NetEQ: fix initial decoder frame length.
Bug: webrtc:10548
Change-Id: If020ce0e5bef57f4f783dbc47995fd0c9f7e2137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161046
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29960}
2019-11-29 13:43:41 +00:00
499b3b6c7e In RtpDepacketizerAV1 use aggregation header to detect key frames
instead of guessing based on presence of the sequence header OBU.

Bug: webrtc:11042
Change-Id: I9a0674249feceebb07299ea965c62e87499f6baf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161013
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29958}
2019-11-29 10:14:20 +00:00
a3cd717bb6 Remove WebRTC-Bwe-CongestionWindowDownlinkDelay.
Bug: webrtc:11143
Change-Id: Iaf89758de7d2a58f6e1c88293f38c5eff1a78583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160787
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29956}
2019-11-28 21:09:12 +00:00
0682ca9a83 Use AV1 packetizer/depacketizer for AV1 bitstreams
Bug: webrtc:11042
Change-Id: Ibf45a99d8016dccbe109d946ac967efa927312e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161011
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29953}
2019-11-28 18:01:10 +00:00
9dc209a23a Add ability to disable detailed error message in RTC_CHECKs
Bug: webrtc:11133
Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:51:00 +00:00
9750e84d7a AEC3:Turning off default downmix in surround alignment
This CL changes the downmixing of the input to the delay estimation
for surround/stereo signals to be off by default.

A kill-switch is also added for enforcing the downmix to be on.

Bug: webrtc:10913
Change-Id: I1030fef593ba56416deeb13b80d2f3812bffb9ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161012
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29951}
2019-11-28 17:44:40 +00:00
2dec496f80 Add directive to make TRACE_EVENT macros optional.
Bug: webrtc:11132
Change-Id: I801994ad262e1acff73e4c20afd7a7343b56268c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160654
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29949}
2019-11-28 15:58:24 +00:00
b8306cc9bb Remove temporary 8-bit H264 HDR fix
Bug: webrtc:10575, chromium:956468
Change-Id: Ie49af9c9624962bd19147833a167e5830bb81fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161004
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29948}
2019-11-28 15:32:04 +00:00
096a46f38f Implement AV1 RtpPacketizer class
Bug: webrtc:11042
Change-Id: Id1fc0acfa87a4520344f2636f50cb4d4e7284829
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160416
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29947}
2019-11-28 14:39:02 +00:00
4314a494cf Implements a task-queue based PacedSender, wires it up for field trials
Bug: webrtc:10809
Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-28 12:13:53 +00:00
5314b13a8d Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame
Bug: chromium:1028348
Change-Id: I824e84138acbf4e73fc21ee8248e29e5cc7a0ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160643
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29945}
2019-11-28 11:27:33 +00:00
5cef9c3581 Revert "Add support for RtpEncodingParameters::max_framerate"
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.

Reason for revert: crbug.com/1028937

Original change's description:
> Add support for RtpEncodingParameters::max_framerate
> 
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
> 
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org

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

Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
2019-11-27 14:01:53 +00:00
9f9e20a3dc Fix errorprone issues preventing Chromium Roll.
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
26cc5e650f Corrected the aggregation of AGC choices and add fallback solution
This CL corrects the analog AGC code so that the levels are properly
aggregated and not only the level of the first channel is chosen.

It also adds a kill-switch to allow the aggrated level to be the maximum
level rather than the minimum level.

Bug: webrtc:10859
Change-Id: Ibf4fecb53cfaf0dc064c334112105bf26401f78d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160708
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29931}
2019-11-27 11:57:22 +00:00
17e4c58318 Adding parametrization of the AEC3 howling mitigation behavior
Bug: webrtc:8671,b/145243047
Change-Id: If5bcbb66b72278b901a990cb9d6e11e42c9ac592
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160781
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29930}
2019-11-27 11:56:17 +00:00
b1ccae253e Reland "Fixes dynamic mode pacing issues."
This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f
Was not the cause of perf alert, relanding.

TBR=ilnik@webrtc.org

Original change's description:
> Fixes dynamic mode pacing issues.
>
> This CL fixes a few issues in the (default-disabled) dynamic pacing
> mode:
> * Slight update to sleep timing to avoid short spin loops
> * Removed support for early execution as that lead to time-travel
>   contradictions that were difficult to solve.
> * Makes sure we schedule a process call when a packet is due to be
>   drained even if the queue is empty, so that padding will start at
>   the correct time.
> * While paused or empty, sleep relative last send time if we send
>   padding while silent - otherwise just relative to last process
>   time.
> * If target send time shifts so far back that packet should have
>   been sent prior to the last process, make sure we don't let the
>   buffer level remain.
> * Update the PacedSender test to _actually_ use dynamic processing
>   when the param says so.
>
> Bug: webrtc:10809
> Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29911}

Bug: webrtc:10809
Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786
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@{#29928}
2019-11-27 10:22:05 +00:00
dc36829db0 Add VideoCodecType::kVideoCodecAV1 value
Bug: webrtc:11042
Change-Id: I3c5151c9e47679760f8f7d79270488fa8f4c7db5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159282
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29927}
2019-11-27 10:18:45 +00:00
e14cb99408 Correct/update the activation of the multi-channel processing in APM
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.

It also splits the activation of multi-channel processing to be separate
for render and capture.


Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
2019-11-27 10:15:25 +00:00
2a6b3b1f7f Correcting the analog AGC re-initialization at device changes
This CL corrects the re-initialization behavior of the analog
AGC to work correctly when the AGC is reinitialized.

Bug: webrtc:11131
Change-Id: Ie455ba3db1aa3936cbcbb2fab023528124853284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160650
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29924}
2019-11-26 20:30:20 +00:00
3967389d34 Revert "Fixes dynamic mode pacing issues."
This reverts commit 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f.

Reason for revert: Speculative revert due to perf change

Original change's description:
> Fixes dynamic mode pacing issues.
> 
> This CL fixes a few issues in the (default-disabled) dynamic pacing
> mode:
> * Slight update to sleep timing to avoid short spin loops
> * Removed support for early execution as that lead to time-travel
>   contradictions that were difficult to solve.
> * Makes sure we schedule a process call when a packet is due to be
>   drained even if the queue is empty, so that padding will start at
>   the correct time.
> * While paused or empty, sleep relative last send time if we send
>   padding while silent - otherwise just relative to last process
>   time.
> * If target send time shifts so far back that packet should have
>   been sent prior to the last process, make sure we don't let the
>   buffer level remain.
> * Update the PacedSender test to _actually_ use dynamic processing
>   when the param says so.
> 
> Bug: webrtc:10809
> Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29911}

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

Change-Id: I5d1532d2e041e60a7f1bfeb8185f7760c9789711
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29920}
2019-11-26 17:02:39 +00:00
68c6572980 Add a CreateNetEq method that takes an AudioDecoderFactory
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.

Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
2019-11-26 14:43:49 +00:00
a88655daf9 NetEQ RTP play: textlog to stderr as option
Bug: webrtc:10548
Change-Id: I260b6c63621c61e33fcc38fd0a39cfb0dba3bc20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160413
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29915}
2019-11-26 11:50:54 +00:00
27bd76bcb2 DCHECKing for deprecated 8kHz support in AGC and changing fuzzer
This CL adds a DCHECK for the deprecated 8 kHz rate in APM.
It also updates the agc fuzzer code to properly do band-split on
the signals, and not send 8 kHz signals into the AGC.

Bug: chromium:1028092,chromium:1028172
Change-Id: I1e7c8d721834310e94b0e21efea07f75da837cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160600
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29914}
2019-11-26 11:10:54 +00:00
a101a4f186 Reland "Add IvfVideoFrameGenerator"
This is a reland of 712a26f3842b4eba1f38c3ba7371b1cf771fd232

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

Bug: webrtc:10138
Change-Id: If522d079f0a1e30d6f2b330792aa1d1fc043b8b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160418
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29913}
2019-11-26 09:56:47 +00:00
72e6cb0b3f Fixes dynamic mode pacing issues.
This CL fixes a few issues in the (default-disabled) dynamic pacing
mode:
* Slight update to sleep timing to avoid short spin loops
* Removed support for early execution as that lead to time-travel
  contradictions that were difficult to solve.
* Makes sure we schedule a process call when a packet is due to be
  drained even if the queue is empty, so that padding will start at
  the correct time.
* While paused or empty, sleep relative last send time if we send
  padding while silent - otherwise just relative to last process
  time.
* If target send time shifts so far back that packet should have
  been sent prior to the last process, make sure we don't let the
  buffer level remain.
* Update the PacedSender test to _actually_ use dynamic processing
  when the param says so.

Bug: webrtc:10809
Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29911}
2019-11-25 23:11:02 +00:00
13ea34f305 Revert "Add IvfVideoFrameGenerator"
This reverts commit 712a26f3842b4eba1f38c3ba7371b1cf771fd232.

Reason for revert: consistently failing on iOS64 Debug: https://ci.chromium.org/p/webrtc-internal/builders/ci/iOS64%20Debug/20119

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

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

Change-Id: Ie34e254a7a4ff5ff8fdab7c6b3212792b52b6f53
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160560
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29910}
2019-11-25 21:58:02 +00:00
15be5282e9 Add support for RtpEncodingParameters::max_framerate
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.

Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
2019-11-25 16:43:59 +00:00
f534a64047 AEC3: Sub-band nearend detector
Implements an alternative to the dominant nearend detector.

Bug: b/130016532
Change-Id: If4867d58aad036ccf4e456ef81689b8db0284f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159865
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29906}
2019-11-25 16:26:49 +00:00
712a26f384 Add IvfVideoFrameGenerator
Bug: webrtc:10138
Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29902}
2019-11-25 13:38:09 +00:00
c421f3ef15 Makes sprang@ owner in modules/pacing
Bug: None
Change-Id: I4eca1d7f3af2fe949b368924e84b7f3d040d22ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160405
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29899}
2019-11-25 12:32:18 +00:00
80b2806250 Fixing a buffer overflow in Merge::Downsample
In the unlikely event that the decoded audio is really short, the
downsampling would read outside of the decoded audio vector. This CL
fixes that, and adds a unit test that verifies the fix (when running
with ASan).

Bug: chromium:1016506
Change-Id: Ifb8071ce0550111cd66e7f7c1bed7f17b33f93c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160304
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29898}
2019-11-25 12:16:30 +00:00
77dc19905d Changed the digital AGC1 gain to properly support multichannel
Beyond making the digital AGC1 code properly support
multichannel, this CL also
-Removes deprecated debug logging code.
-Converts the gain application to be fully in floating point
 which
--Is less computationally complex.
--Does not quantize the samples to 16 bit before applying the
  gains.

Bug: webrtc:10859
Change-Id: I6020ba8ae7e311dfc93a72783a2bb68d935f90c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29886}
2019-11-23 08:42:59 +00:00
e35b32c29f AGC: Removing unnneccessary copying and changing to using const
The changes have been shown to be bitexact on a large dataset.

Bug: webrtc:10859
Change-Id: Iedc0e9e944ebfabb717dd7fb4d2682c695da883e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159694
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29883}
2019-11-22 20:14:16 +00:00
9cb06610d2 Add multi-channel support to AECM
AECM only supports up to two capture channels, this CL extends it to arbitrary channel counts.

Bug: webrtc:10859
Change-Id: Id56ca633cd9de706fa1254bfa8153de88de0ef70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160340
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29880}
2019-11-22 17:16:53 +00:00
3daedb6c88 Making the Analog AGC properly support multi-channel
This CL adds proper multi-channel support to the analog AGC.

Beyond that, it prepares adding multi-channel support to the digital
AGC by removing the tight dependency between the analog and digital
AGC codes.

Bug: webrtc:10859
Change-Id: I4414ccbc3db5dbb5ae069fdf426cbd038375ca7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159480
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29878}
2019-11-22 15:32:55 +00:00
af070d0299 Improves device enumeration in ADM2 for Windows.
Summary of changes/improvements and fixes:

Changes container for list of devices from std::vector to std:deque to
allow fast insertion and deletion at both its beginning and its end. This
approach makes it easier to first build a list of all available devices
and then check the size of the list. If size > 0 => two more devices are
added at the front (Default and Default Communication). The old solution
contained a risk of adding invalid Default and Default Communication
devices in cases where not physical device could be found.

Adds usage of |device_index_| in CoreAudioBase to ensure that the selected
device is unique. The previous version used only an ID but that ID is not
unique when e.g. only one device exists since it can have up to three
different roles.

Improves logging and comments.

No-Try: True
Tbr: thaloun@chromium.org
Bug: webrtc:11107
Change-Id: I9a09f7716ed8d8858dcc6a5354b038fc06496166
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160050
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29874}
2019-11-22 14:27:10 +00:00
9281436650 Add field trial to cap trendline slope in delay-based BWE.
Bug: webrtc:10932
Change-Id: I34a36a8cad16d65143eff9c675ee98bdbf176ace
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160014
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29872}
2019-11-22 13:14:53 +00:00
efbda8d90a Don't perform DataCallback if the input object has been stopped.
Fix signed/unsigned mismatch.


Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results.

It's possible for an device to be counted but getting its name fails, in which case the utility function returns true but would continue from its loop filling the AudioDeviceNames vector, leading to a smaller output than the later code expects.

Bug: b/144382120
Change-Id: Iab008c28f03023c830011d229b1f1c7e3e7bb5ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160226
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29871}
2019-11-22 10:18:39 +00:00
4995f872ca Cleans up the round robin packet queue.
Usage of this class has now been simplified so that we can do some
cleanup:

* Removes dead code: Push() with 9 args, CancelPop()
* Replaces BeginPop()/CancelPop() with a single Pop() method
* Makes QueuePacket a private class
* Replaces rtp_packets_ with direct ownership from QueuePacket

Bug: webrtc:10809
Change-Id: Iea131ee87d5d920360c71fb180b2af0ea4fc6c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160007
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29869}
2019-11-22 08:07:15 +00:00
0174ffe700 Can list UWP applications
Before the CL [1] https://webrtc-review.googlesource.com/c/src/+/144960
the UWP applications could not be captured so they were filtered out.

Another reason of this previous filter was because otherwise some
'ghost windows' are listed too. These 'ghost windows' are prelaunched
UWP apps whose windows are created in a hidden/cloaked state to improve
perceived performance of launching these apps later, see:
[2] https://docs.microsoft.com/en-us/windows/uwp/launch-resume/handle-app-prelaunch

They can be filtered out using the new API merged recently
'webrtc::WindowCaptureHelperWin::IsWindowCloaked, see:
[3] https://webrtc-review.googlesource.com/c/src/+/143980

This patch allows non-cloaked UWP apps to be listed by taking advantage
of CL [3]. So that user can select them with the app window picker and can
then share them thanks to [1].

Bug: chromium:700037
Change-Id: I4b41bb764ebbd6e2f164f036a63a4b1cd06c3f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160021
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29868}
2019-11-21 23:35:09 +00:00
b0df593e56 Reland "Prepares PacingController for simplified packet queue."
This is a reland of acdc22d7845c5dde7c23366110e54e5d26127c85

Original change's description:
> Prepares PacingController for simplified packet queue.
>
> This CL removes references to RoundRobinPacketQueue::QueuedPacket,
> other than the method to release an RtpPacketToSend. It also moves
> both the BeginPop() and FinalizePop() to within a single helper
> method.
>
> A follow-up cleanup of the packet queue will stop exposing the
> QueuedPacket struct and replaces the the pop-methods with a single
> new one that just returns an RtpPacketToSend.
>
> Bug: webrtc:10809
> Change-Id: I5208a93e12e6b56714d483cc12d2a37225ea8e5e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159889
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29820}

TBR=philipel@webrtc.org

Bug: webrtc:10809
Change-Id: Id8196d9348d7fa69a5e410367b8a88e6039ef1b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160205
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29867}
2019-11-21 17:19:10 +00:00
58a3210823 Add config to reduce weight on small samples in BitrateEstimator.
Change #159711 adds the option to filter out small packets on the
input to the delay-based BWE. This change adds similar functionality
to BitrateEstimator by reducing the weight of small observations.

Bug: webrtc:10932
Change-Id: I0a673a067f7ef86769cabd30443e60e9de70053c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160009
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29865}
2019-11-21 15:52:25 +00:00