Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast.
Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000
Hiding snprintf definition if building with Visual Studio 2015
Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc.
BUG=webrtc:5183
Review URL: https://codereview.webrtc.org/1412653006
Cr-Commit-Position: refs/heads/master@{#11434}
The level of the error signal after linear echo cancellation was based on non-buffered signal while that of the near-end and far-end signal based on buffered signal. This discrepancy made the comparison of them unfair.
This CL is to make calculating the error level rely on the same buffering.
BUG=
Review URL: https://codereview.webrtc.org/1510873004
Cr-Commit-Position: refs/heads/master@{#11408}
In AEC, audio levels are calculated in frequency domain. This makes the calculation dependent on FFT. We now make the calculation performed in time domain. The complexity is the same, but the dependence on FFT is removed.
BUG=
Review URL: https://codereview.webrtc.org/1542573002
Cr-Commit-Position: refs/heads/master@{#11357}
Reason for revert:
Reverting due to problem with roll:
/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
-> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
configs -= [ "//build/config/clang:find_bad_constructs" ]
^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@
Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}
TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1586563003
Cr-Commit-Position: refs/heads/master@{#11239}
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Review URL: https://codereview.webrtc.org/1538643004
Cr-Commit-Position: refs/heads/master@{#11231}
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
Currently, FFT is performance when AEC buffers farend signal. This has some drawbacks
1. memory inefficiency: two ring buffers are needed;
2. computation inefficiency: if ringbuffer gets wrapped around, some FFT computation will be wasted;
3. accessibility: the main AEC function looses accessibility to the time-domain signal.
Therefore, this CL tries to buffer time domain data, which is buffered any way if a debugging macro is defined, and calculate the FFTs where they are actually used.
BUG=
Review URL: https://codereview.webrtc.org/1512573003
Cr-Commit-Position: refs/heads/master@{#11091}
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
-Extended the InverseFft function to be more generally
applicable.
-Included the previous external extra scaling into the
preexisting InverseFft call.
-Moved the updating of aec->delayEstCtr to where it is
actually used.
-Refactored the output production and comfort noise
addition using the InverseFft function.
-Removed the if-statements checking the value of the
constant flagHbandCn as any value different from 1 would
crash the program. Also removed the constant
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1492343002
Cr-Commit-Position: refs/heads/master@{#11054}
-Changed the type for the frequency estimate of the comfort noise for the
higher band to be a two dimensional float array instead of a complex_t array.
This makes sense since all the other frequency estimate (apart from the
coherence) use this format and doing this change allows bundling the
IFFT operations into using the InverseFFT method.
-Moved the memset of the frequency estimate of the comfort noise to where it is used and made it conditional so that it is only performed when used.
-Harmonized the if-statements for when the frequency estimate of the comfort noise is computed in the different optimized ComfortNoise computation methods.
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494133002
Cr-Commit-Position: refs/heads/master@{#11050}
using the wrong sample rate for the render signal.
The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.
The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).
It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1525173002
Cr-Commit-Position: refs/heads/master@{#11045}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
-Moved memsets to where their variables are used.
-Removed redundant.
-Changed a pointer scalar to be accessed in pointer notation rather than
in array notation.
The change has been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494473006
Cr-Commit-Position: refs/heads/master@{#10963}
-Moved filter reset from the echo suppression
into the echo subtraction code where it belongs
(the echo subtractor should own its filter reset).
-Moved the selection between using the microphone sinal and
the echo subtractor output down to the lowest level in the
EchoSuppression function. This makes sense as that selection
was very hidden in an unrelated sub-sub-function call and
as the selection is critical for what the AEC outputs.
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1499573003
Cr-Commit-Position: refs/heads/master@{#10956}
to simplify future refactoring and development.
In more detail:
1) Moved the updating of eBuf from the EchoSubtraction method
to the EchoSuppression method as it is only used in the latter.
2) Moved the computation of efw and dfw from the SubbandCoherence method
as those are actually the analysis filterbank computation that is not
directly related to the coherence.
3) As a consequence of 2) 3 functions needed to be replaced by the
generic function pointer scheme used in WebRTCAec as they have
optimized versions for SSE2 and NEON (which before were local to each
of the aec_core*.c files.
Motivation:
Apart from making sense from a logical point of view, the changes will
a) Allow eBuf stored in half the size on the state.
b) Allow simpler switching between using the the microphone signal
and echo subtractor output in the echo suppressor.
c) Allow further refactoring that move all the changes to eBuf to one method
(currently those are happening in at least 4 different methods.
Drawbacks:
i) dfw is moved to EchoSuppression which increases the stack usage for that
method. This will, however, be improved once further refactoring can be done.
The changes have been tested for bitexactness on Linux using a quite extensive dataset.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494563002
Cr-Commit-Position: refs/heads/master@{#10954}
To try to resolve the problem I replaced the custom synchronization with rtc::Event which made the code cleaner, faster, and less error prone.
However, in the end the source of the test locks was that during TearDown one of the threads was stuck in a waiting loop.
I added a fix for the TearDown issue but still decided to keep the rtc:Event - based code change metioned above as that gave a more clean code.
BUG=
Review URL: https://codereview.webrtc.org/1490113004
Cr-Commit-Position: refs/heads/master@{#10880}
The two added macros simplifies the logging code when a value which is not stored in a variable should be logged.
BUG=
Review URL: https://codereview.webrtc.org/1488613002
Cr-Commit-Position: refs/heads/master@{#10870}
-Renamed the TimeToFrequency and FrequencyToTime functions.
-Moved the windowing from the TimeToFrequency function.
-Simplified the EchoSubtraction function.
Note that the aec state is still an input to the EchoSubtraction function, and it currently needs to be that in order to support the output of the debug file. The longer-term goal is, however, to order the state into substates. This will simplify the parameter lists to the EchoCancellation function as well as replace the aec state as a parameter
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1456123003
Cr-Commit-Position: refs/heads/master@{#10830}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Reason for revert:
This breaks the Win32 Release [large tests] bot (webrtc_perf_tests times out after 1h23m): https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D
The Mac64 Release [large tests] bot's runtime also increased with +20 minutes.
These bot configs are not a part of the default trybot set, so please run them manually or add this to the CL description:
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
Original issue's description:
> A unittest that reports the statistics for the duration of an APM stream processing API call.
>
> BUG=webrtc:5099
>
> Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb
> Cr-Commit-Position: refs/heads/master@{#10786}
TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org,peah@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1473733004
Cr-Commit-Position: refs/heads/master@{#10791}
This works aims to:
-More clearly separate the functionalities in the AEC.
-Make the inputs and outputs to functions more clear (currently the state struct is often passed as a parameter to the functions and the functions use members of the state both as inputs and outputs, which reduces the readability of the code and makes it difficult to change/refactor.
What is done in this CL:
-Most of what belongs to the echo subtraction functionality has been moved to a separate function.
-The NonLinearProcessing function has been renamed to EchoSuppressor which I think is more appropriate.
-Part of the code was replaced by a call to the TimeToFrequency function (which was also suggested by an existing todo).
-For consistency, a function FrequencyToTime doing the opposite of TimeToFrequency was added and part of the code was moved to that.
-The ScaleErrorSignal function was changed to no longer have the state as an input parameter. This entailed also changing the corresponding assembly optimized files accordingly.
Testing:
-The changes have been tested for bitexactness on Linux using a fairly extensive test.
-All the unittests pass on linux.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1455163006
Cr-Commit-Position: refs/heads/master@{#10764}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
