Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
- Remove webrtc/tools/agc/test_utils.cc/.h - only used from the above test.
- Remove webrtc/tools/agc/agc_harness.cc - not used anymore.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2299023004
Cr-Commit-Position: refs/heads/master@{#14039}
Currently, the aec_debug_dump buildflag can and is used to store data in the whole of
the audio processing module. Therefore a more appropriate name is apm_debug_dump which
also matches the names of the data dumping functionality. This CL makes that name change.
The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to
WEBRTC_APM_DEBUG_DUMP == 1
Furthermore, this CL moves the buildflag to a more appropriate place.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2300813004
Cr-Commit-Position: refs/heads/master@{#14026}
when building the code for ARM.
The intention is to follow up this CL with other CLs that
further addresses the internal resampling in APM
BUG=webrtc:6181
Review-Url: https://codereview.webrtc.org/2265473003
Cr-Commit-Position: refs/heads/master@{#13974}
Reason for revert:
This caused build breakage due to upstream dependencies.
These dependencies need to be resolved before landing the CL.
Original issue's description:
> This CL adds functionality in the level controller to
> receive a signal level to use initially, instead of the
> default initial signal level.
>
> BUG=
>
> Committed: https://crrev.com/57fec1d828113241186e78710ec5e851cc1a0e81
> Cr-Commit-Position: refs/heads/master@{#13931}
TBR=henrik.lundin@webrtc.org,aleloi@webrtc.org,solenberg@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2283793002
Cr-Commit-Position: refs/heads/master@{#13936}
receive a signal level to use initially, instead of the
default initial signal level.
BUG=
Review-Url: https://codereview.webrtc.org/2254973003
Cr-Commit-Position: refs/heads/master@{#13931}
We have RTC_CHECK and RTC_DCHECK for C now, so we should use it. It's
one fewer difference between our C and C++ code.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2274083002
Cr-Commit-Position: refs/heads/master@{#13930}
I saw this when browsing the code, I think the intended behavior is accumulating to a float.
BUG=none
Review-Url: https://codereview.webrtc.org/2268163004
Cr-Commit-Position: refs/heads/master@{#13918}
Reason for revert:
Breaks most of chromium.webrtc.fyi bots.
Original issue's description:
> GN build rules for four audio processing test executables
>
> click_annotate, intelligibility_proc, nonlinear_beamformer_test, and
> transient_suppression_test.
>
> BUG=webrtc:5949
>
> Committed: https://crrev.com/538b5606a3fb6310aab7a7e747aee16eac885f02
> Cr-Commit-Position: refs/heads/master@{#13890}
TBR=kjellander@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2274813004
Cr-Commit-Position: refs/heads/master@{#13891}
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.
BUG=webrtc:6215
NOTRY=True
Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
bitexactness test. The activation of the test will
be done in another CL.
BUG=
Review-Url: https://codereview.webrtc.org/2257733002
Cr-Commit-Position: refs/heads/master@{#13822}
Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"
Removed audio_buffer_tools from modules_unittests to match the gyp file.
BUG=webrtc:6041
Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity.
R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/2053773002 .
Cr-Commit-Position: refs/heads/master@{#13352}
It does:
-Handle saturations in a better manner by adding different gain change
step sizes for upwards and downwards changes, as well as when there
is saturation.
-Handle conditions with initial noise-only regions in a better way by
setting a high initial peak level estimate which is gradually reduced until
certainty about the peak level is achieved.
-Limit the maximum gain to limit noise amplification, and to reflect that it
initially is intended to be used in cascade with the fixed digital AGC mode.
-Lower the maximum allowed stationary noise floor to reduce the risk of
excessive noise amplification.
-Lower the target gain to reduce the risk of causing the AEC on the other
end to fail due to high playout levels triggering nonlinearities.
This also reduces the risk for saturation.
-Handle the noise-only regions in a better manner.
NOTRY=true
TBR=aleloi
BUG=webrtc:5920
Review-Url: https://codereview.webrtc.org/2111553002
Cr-Commit-Position: refs/heads/master@{#13350}
Label less chunks as speech, adapt slower and be more conservative with the maximum gain it can apply.
Review-Url: https://codereview.webrtc.org/2087623003
Cr-Commit-Position: refs/heads/master@{#13242}
Also added IntelligibilityEnhancer setting to aecdump simulator in audioproc_f
Review-Url: https://codereview.webrtc.org/2075093003
Cr-Commit-Position: refs/heads/master@{#13220}
Some aecdumps have more than one INIT event. In those cases only the last wav file was unpacked, which sometimes is not the most interesting or desired one.
This CL creates a different wav file after each INIT event.
Review-Url: https://codereview.webrtc.org/2067423002
Cr-Commit-Position: refs/heads/master@{#13196}
This is a somewhat involved refactoring of this class. Here's an overview of the changes:
* FileWrapper can now be used as a regular class and instances allocated on the stack.
* The type now has support for move semantics and copy isn't allowed.
* New public ctor with FILE* that can be used instead of OpenFromFileHandle.
* New static Open() method. The intent of this is to allow opening a file and getting back a FileWrapper instance. Using this method instead of Create(), will allow us in the future to make the FILE* member pointer, to be const and simplify threading (get rid of the lock).
* Rename the Open() method to is_open() and make it inline.
* The FileWrapper interface is no longer a pure virtual interface. There's only one implementation so there's no need to go through a vtable for everything.
* Functionality offered by the class, is now reduced. No support for looping (not clear if that was actually useful to users of that flag), no need to implement the 'read_only_' functionality in the class, since file APIs implement that already, no support for *not* managing the file handle (this wasn't used). OpenFromFileHandle always "manages" the file.
* Delete the unused WriteText() method and don't support opening files in text mode. Text mode is only different on Windows and on Windows it translates \n to \r\n, which means that files such as log files, could have a slightly different format on Windows than other platforms. Besides, tools on Windows can handle UNIX line endings.
* Remove FileName(), change Trace code to manage its own path.
* Rename id_ member variable to file_.
* Removed the open_ member variable since the same functionality can be gotten from just checking the file pointer.
* Don't call CloseFile inside of Write. Write shouldn't be changing the state of the class beyond just attempting to write.
* Remove concept of looping from FileWrapper and never close inside of Read()
* Changed stream base classes to inherit from a common base class instead of both defining the Rewind method. Ultimately, Id' like to remove these interfaces and just have FileWrapper.
* Remove read_only param from OpenFromFileHandle
* Renamed size_in_bytes_ to position_, since it gets set to 0 when Rewind() is called (and the size actually does not change).
* Switch out rw lock for CriticalSection. The r/w lock was only used for reading when checking the open_ flag.
BUG=
Review-Url: https://codereview.webrtc.org/2054373002
Cr-Commit-Position: refs/heads/master@{#13155}
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
This reduces its complexity by a factor of 2.7x total.
The mean error introduced by this is in the 6 different noise scenarios and 6 different speech signals tested is below -52dB.
Review-Url: https://codereview.webrtc.org/2035213002
Cr-Commit-Position: refs/heads/master@{#13072}
The changes are done in several patches in order to make
the review easier.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2051443002
Cr-Commit-Position: refs/heads/master@{#13068}
