There were a few typos in the README file.
Bug: chromium:362397579
Change-Id: Ib0aa84f57f3d83851f085e595ffa72a53ec8311d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43525}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
Also CHECK in OutputPathWithRandomDirectory. This function is used in tests that need a unique folder to avoid interaction with other tests that may run in parallel. Continuing with a non-unique folder if the creation fails, is likely to cause surprising errors later on.
Bug: webrtc:379973428
Change-Id: I6a30ef9034be8132e2362eff5e46e3b99b30acd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43431}
This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.
Instead of InitializeIfNeeded:
* Offer a way to explicitly initialize PushResampler via the ctor
(needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
(All calls to Resample() were preceded by a call to Initialize)
As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.
Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.
Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
Allow skipping the deinterleaving steps in PushResampler
before resampling when deinterleaved buffers already exist.
Bug: chromium:335805780
Change-Id: I2080ce2624636cb743beef78f6f08887db01120f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352202
Reviewed-by: Per Åhgren <peah@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42438}
PushResampler now uses a single buffer for the deinterleaved
channel sources and another for the deinterleaved destinations.
Before, there was a dedicated buffer per channel (source and dest).
This reduces allocations and allows for using DeinterleavedView for
both which simplifies some checks.
Bug: chromium:335805780
Change-Id: I553a36164109127fa332ab17918d53832d442303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351542
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42415}
Interleave and Deinterleave now accept two parameters, one for the
interleaved buffer and another for the deinterleaved one.
The previous versions of the functions still need to exist for test
code that uses ChannelBuffer.
Bug: chromium:335805780
Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42412}
From the new header file:
* MonoView<>: A single channel contiguous buffer of samples.
* InterleavedView<>: Channel samples are interleaved (side-by-side) in
the buffer. A single channel InterleavedView<> is the same thing as a
MonoView<>
* DeinterleavedView<>: Each channel's samples are contiguous within the
buffer. Channels can be enumerated and accessing the
individual channel data is done via MonoView<>.
There are also a few utility functions that offer a unified way to check
the properties regardless of what view type is in use.
Bug: chromium:335805780
Change-Id: I28196f8f4ded4fadc72ee32b62af304c62f4fc47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349300
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42377}
Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
First commit in a series of commits to wire up the test output path utility that adds a random directory in the path, for problematic tests that run in concurrent execution environments.
Bug: webrtc:15833
Change-Id: I5e5b3940007be773d77dbbfc953efa810e4e3ea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339522
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41738}
Providing unique identifiers for files and directories created as part
of unit tests.
Bug: webrtc:15833
Change-Id: If2835c362c47a111aa99b0e3c6ad6a33be061978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41704}
Use signed arithmetic for indexes that can be negative
Bug: webrtc:15523
Change-Id: I41ea47a679242c8febe70373529173733058c479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322120
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40924}
Trying to calculate std::abs(min_int) results in an int overflow.
Original author: asmok@.
Bug: None
Change-Id: I984e9ba4f48411a583a55cc3f9c66c9a1cc8dc92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318120
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40660}
This CL adds the Shipped field (and may update the
License File field) in Chromium READMEs. Changes were
automatically created, so if you disagree with any of
them (e.g. a package is used only for testing purposes
and is not shipped), comment the suggested change and
why.
See the LSC doc at go/lsc-chrome-metadata.
Bug: b:285450740
Change-Id: If4955c6f6e7b58e0c99469fc45ed5b9e8f30a32b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311720
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Anne Redulla <aredulla@google.com>
Cr-Commit-Position: refs/heads/main@{#40424}
This is a port of crrev.com/c/2936677.
Previously we only checked avx2 support and then use avx2/fma
intrinsics in SincResampler(crrev.com/c/2654647),this CL also
checks the fma support and avoids using avx2 code if fma is not
supported.
Bug: chromium:1410691
Change-Id: Ibf7c0a1bead87ebe5d3978cfd20cc23525169f40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291702
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39238}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
* Structs with user-declared constructors are no longer considered
aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
function call" warning
* Various types of math involving enums are deprecated, so replace with
constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
__has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
std::invoke_result(f, x)
Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
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}
