| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| 25702cb1628941427fa55e528f53483f239ae011 |
|
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
Misc. small cleanups.
* 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}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
|
17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
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}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| b0ad43baa02f41dba01be4df9606dc65f24c0ec8 |
|
20-Nov-2015 |
aluebs <aluebs@webrtc.org> |
Add aecdump support to audioproc_f
Originally landed here: https://codereview.webrtc.org/1409943002/
The transient suppression fix landed here: https://codereview.webrtc.org/1411423010/
TBR=mflodman
Review URL: https://codereview.webrtc.org/1432843002
Cr-Commit-Position: refs/heads/master@{#10722}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| b7a5c16d2c6dbe5ca17fca86a3180b8aad5054f7 |
|
05-Nov-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1412963007
Cr-Commit-Position: refs/heads/master@{#10532}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| 86b40506b3443d5cf0c5ec838e44edd9f4376c01 |
|
05-Nov-2015 |
kjellander <kjellander@webrtc.org> |
Reland of Add aecdump support to audioproc_f. (patchset #2 id:250001 of https://codereview.webrtc.org/1423693008/ )
Reason for revert:
Oh dear, this broke compilation.
I guess more was built on top of this CL before I reverted it.
Reverting now for futher investigation (and re-land using CQ)
Original issue's description:
> Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
>
> Reason for revert:
> This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
> I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
>
> See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
>
> Original issue's description:
> > Add aecdump support to audioproc_f.
> >
> > Add a new interface to abstract away file operations. This CL temporarily
> > removes support for dumping the output of reverse streams. It will be easy to
> > restore in the new framework, although we may decide to only allow it with
> > the aecdump format.
> >
> > We also now require the user to specify the output format, rather than
> > defaulting to the input format.
> >
> > TEST=Bit-exact output to the previous audioproc_f version using an input wav
> > file, and to the legacy audioproc using an aecdump file.
> >
> > Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> > Cr-Commit-Position: refs/heads/master@{#10460}
>
> TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/d279941bb54bfdc6e7324bf36cac76581474b96d
> Cr-Commit-Position: refs/heads/master@{#10523}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1419953010
Cr-Commit-Position: refs/heads/master@{#10524}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| d279941bb54bfdc6e7324bf36cac76581474b96d |
|
05-Nov-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1423693008
Cr-Commit-Position: refs/heads/master@{#10523}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| bdafe31b86e9819b0adb9041f87e6194b7422b08 |
|
30-Oct-2015 |
andrew <andrew@webrtc.org> |
Add aecdump support to audioproc_f.
Add a new interface to abstract away file operations. This CL temporarily
removes support for dumping the output of reverse streams. It will be easy to
restore in the new framework, although we may decide to only allow it with
the aecdump format.
We also now require the user to specify the output format, rather than
defaulting to the input format.
TEST=Bit-exact output to the previous audioproc_f version using an input wav
file, and to the legacy audioproc using an aecdump file.
Review URL: https://codereview.webrtc.org/1409943002
Cr-Commit-Position: refs/heads/master@{#10460}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|
| cb05b72eb2f7db4478b93b16faf31ec74237453e |
|
08-May-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Add WAV and arbitrary geometry support to nlbf test.
This adds functionality from audioproc_float. The geometry parsing code
is now shared from test_utils.h. I removed the "mic_spacing" flag from
audioproc_float because it's a redundancy that I suspect isn't very
useful.
Includes a cleanup of the audio_processing test utils. They're now
packaged in targets, with the protobuf-using ones split out to avoid
requiring users to depend on protobufs.
pcm_utils is no longer needed and removed.
The primary motivation for this CL is that AudioProcessing currently
doesn't support more than two channels and we'd like a way to pass
more channels to the beamformer.
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/50899004
Cr-Commit-Position: refs/heads/master@{#9157}
/external/webrtc/webrtc/modules/audio_processing/test/test_utils.cc
|