Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This CL adds functionality that allows adjusting the audio levels
internally in APM. The main purpose of the functionality is to allow
APM to optionally be moved to an integration that does not provide an
analog gain to control, and the implementation of this has been
tailored specifically to meet the requirements for that.
More specifically, this CL does
-Add a new variant of the pre-amplifier gain that is intended to replace
the pre-amplifier gain (but at the moment can coexist with that). The
main differences with the pre-amplifier gain is that an attenuating
gain is allowed, the gain is applied jointly with any emulated analog
gain, and that its packaging fits better with the post gain.
-Add an emulation of an analog microphone gain. The emulation is
designed to match the analog mic gain functionality in Chrome OS (which
is digital) but should be usable also on other platforms.
-Add a post-gain which is applied after all processing has been applied.
The purpose of this gain is for it to work well with the integration
in ChromeOS, and be used to compensate for the offset that there is
applied on some USB audio devices.
Bug: b/177830918
Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33466}
This is a reland of 8be2f201ba
Original change's description:
> Add ability to state whether the APM output will be used
>
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
>
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}
Bug: b/154437967
Bug: b/163802450
Change-Id: Ia77a9e43f913929d1afa72212f1ea6c192d0e519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181887
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31957}
This reverts commit 8be2f201ba.
Reason for revert: Breaks downstream
Original change's description:
> Add ability to state whether the APM output will be used
>
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
>
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}
TBR=alessiob@webrtc.org,peah@webrtc.org
Change-Id: I1e56dafbbfa6ea69cccbbb5cdc2b1e2a6c122c11
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/154437967
Bug: b/163802450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181884
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31953}
This CL adds the ability for the surrounding code to state that the
APM output will not be used. The intended usecase for this is to allow
APM to run at a lower complexity when the endpoint is muted.
When APM has been informed that the output will not be used, it can
turn off code that is needed only for ensuring that the output audio
will sound good.
Bug: b/154437967,b/163802450
Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31949}
This CL creates a new stream interface and uses it to replace
most of the usage of AudioFrame in the non-test code.
The CL changes some of the test code as well, as the other
changes required that.
The CL will be followed by 2 more related CLs.
Bug: webrtc:5298
Change-Id: I5cfbe6079f30fc3fbf35b35fd077b6fb49c7def0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170040
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30799}
Add a runtime setting that notifies play-out audio device changes.
The payload is a pair indicating a device id and its maximum play-out
volume.
kPlayoutVolumeChange is now forwarded not only to capture, but also
render (required by render_pre_processor).
Bug: webrtc:10608
Change-Id: I8997c207422c1dcd1d53775397d6290939ef3db8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159002
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29725}
Add a PlayoutVolumeChange RuntimeSetting. Trigger an echo path change when the playout volume is changed.
Bug: webrtc:10608
Change-Id: I1e736b93c1865d08c7d2582f6fe00216c1e1f72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135746
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27913}
This CL extends the supported runtime settings in
APM to also comprise the AGC2 fixed gain.
The CL was originally created by Adam Whiteside.
Bug: webrtc:10574
Change-Id: I79b3d6501f1e202b66a9b6018f8a493a56b01f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27782}
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
GainControlImpl into the GainControlConfigProxy, as it becomes the
sole AGC object with functionality exposed to the client.
Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
This is a reland of 80b95de765
Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
>
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}
Bug: webrtc:6463
Change-Id: I12154ef65744c1b7811974a1d871e05ed3fbbc27
Reviewed-on: https://webrtc-review.googlesource.com/c/118660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26337}
Also read and apply settings when parsing and replaying dumps.
The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
audioproc_f.
Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502
':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.
Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.
NOTRY=true # want this in and 'ios_api_framework' seems stuck.
Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.
Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.
Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
Between patch set 4 and patch set 5 in
https://codereview.webrtc.org/2865113002/, a line consisting of a
single 'std::move(task);' was added. The reason we will never know,
because the author will not tell. The superfluous line would have gone
unnoticed except for occasional raised eyebrows of casual code
readers.
The Visual Studio compiler now sees lines that have no effect. Which
was announced to the world in the tweet
https://twitter.com/StephanTLavavej/status/924011366943354880
achieving 27 likes and 6 retweets.
Bug: webrtc:8463
Change-Id: Iac49bc4153254b6cfe99f609da28eb4f43ff765e
Reviewed-on: https://webrtc-review.googlesource.com/21324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20616}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc (Browse further)
