| 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/process_test.cc
|
| a4df27b6713583045e51e20c4eb93718d15ca33e |
|
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
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}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| f4f5cb09277d5ef6aeac8341e5f54a055867803a |
|
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
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
Review URL: https://codereview.webrtc.org/1541633002
Cr-Commit-Position: refs/heads/master@{#11093}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 36d4c545007129446e551c45c17b25377dce89a4 |
|
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
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}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| ae2c5ad12afc8cc29fe9c59dea432b697b871a87 |
|
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
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
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.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/process_test.cc
|
| c1cd2bbd790bcf0c6b3ef8abdd5d2a9bca8d10e7 |
|
09-Nov-2015 |
peah <peah@webrtc.org> |
Turned off progress report for finished processing when the progress report is explicitly deactivated
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1407723002
Cr-Commit-Position: refs/heads/master@{#10566}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.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/process_test.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/process_test.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/process_test.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.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/process_test.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 0f133b99c655cbdb347b4a71ac872c071532189f |
|
02-Jul-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Rename APM Config ReportedDelay to DelayAgnostic
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 441f6347311bcf2079435c3888d67e1fb321f9f8 |
|
09-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec."
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 3fbf3f8841b5460503fb646eaedcb063620434a8 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.
BUG=4696
TBR=bjornv@webrtc.org
Review URL: https://codereview.webrtc.org/1166463006
Cr-Commit-Position: refs/heads/master@{#9380}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.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/process_test.cc
|
| beb9798ab4c334617e3023a9c2811e1e34b5a49f |
|
28-Apr-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool
The way SetExtraOptions() is used today only applies for any one configuration change. The correct way is to set it after all flags have been scanned.
The prefered way to solve this is to use gflags and scan once, followed by applying the configuration when creating audio_processing. This is what is done in the new test tool audioproc_float.cc, but there are still some things left to do before we can replace this one.
BUG=N/A
TESTED=locally
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45279004
Cr-Commit-Position: refs/heads/master@{#9097}
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| d35a5c350617cc9d60ce45201764a99229b7299a |
|
10-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make ChannelBuffer aware of frequency bands
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| b1786dbab00dd66a9e59a68414e85b2b2615a24f |
|
03-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
To more easily determine if for example the AEC is not working properly one could monitor how often the estimated delay is out of bounds. With out of bounds we mean either being negative or too large, where both cases will break the AEC.
A new delay metric is added telling the user how often poor delay values were estimated. This is measured in percentage since last time the metrics were calculated.
All APIs have been updated with a third parameter with EchoCancellation::GetDelayMetrics() giving the option to exclude the new metric not to break existing code.
The new metric has been added to audio_processing_unittests with an additional protobuf member, and reference files accordingly updated.
voe_auto_test has not been updated to display the new metric.
BUG=4246
TESTED=audioproc on files
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39739004
Cr-Commit-Position: refs/heads/master@{#8230}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8230 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| ee30082af83c185a121b0055be8ed90a39d3d3e3 |
|
13-Nov-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Set correct sample rate in far_frame in audioproc tool.
One debug recording with non matching sample rates between render and capture revealed a bug in modules/audio_processing/test/process_test.cc
The far_frame (render audio frame) used was loaded with the capture rate instead of the render rate with a data length mismatch error as result.
BUG=N/A
TESTED=manually on linux
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7695 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| a3ed713dad5ccad03e2f5d775081143babd19097 |
|
31-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a WavReader counterpart to WavWriter.
Don't bother with a C interface as we currently have no need to call
this from C code. The first use will be in the audioproc tool.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 634c926928f60a819c23adc3781a49592a1305ef |
|
24-Sep-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audioproc: Now also writes to output file in simulation mode
After changing to use wav as default file format no output was written in simulation mode.
BUG=3359
TESTED=locally
R=aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7286 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 021e76fd39f1cc6500b398b27778c7a814463bfd |
|
04-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Add support for WAV output in audioproc
The default output is a WAV file, except if the --pcm_output flag is set.
BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 74cf9169240ba6867f7b9f210507c146b37da522 |
|
03-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Fix issues in audioproc for float aecdumps
* The right buffer size is used to dump to file when the output sample rate is different from the input one.
* The percentage of processed chunks is calculated correctly when float data available.
BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7036 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 9825afc3bd9759c452e55e97966129e0c6a8a9ed |
|
30-Jun-2014 |
aluebs@webrtc.org <aluebs@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add ExperimentalNs support in Config
R=andrew@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 84f8ec1f9ca61c54de4f56c67f3db495feab1e38 |
|
19-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Changes to tests and tools in audio_processing.
- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.
- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.
BUG=N/A
R=aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 8f69330310bf786cff373c225967e7459fb0b560 |
|
26-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace scoped_array<T> with scoped_ptr<T[]>.
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| ddbb8a2c243f9d54cb0ce0092e341dfc6e126bb3 |
|
22-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| b13a7d5b1c6a30a6f662a7f2153f6ccbfe3c1c34 |
|
27-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Don't disable experimental AGC in audioproc.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5798 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| a8b97373d5d3154357cc6589ff949ee9f6f99d8d |
|
10-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add tests and modify tools for new float deinterleaved interface.
- Add an Initialize() overload to allow specification of format
parameters. This is mainly useful for testing, but could be used in
the cases where a consumer knows the format before the streams arrive.
- Add a reverse_sample_rate_hz_ parameter to prepare for mismatched
capture and render rates. There is no functional change as it is
currently constrained to match the capture rate.
- Fix a bug in the float dump: we need to use add_ rather than set_.
- Add a debug dump test for both int and float interfaces.
- Enable unpacking of float dumps.
- Enable audioproc to read float dumps.
- Move more shared functionality to test_utils.h, and generally tidy up
a bit by consolidating repeated code.
BUG=2894
TESTED=Verified that the output produced by the float debug dump test is
correct. Processed the resulting debug dump file with audioproc and
ensured that we get identical output. (This is crucial, as we need to
be able to exactly reproduce online results offline.)
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| bc1d22461b300624a8b0f81c1535624f4cd6e95a |
|
25-Feb-2014 |
aluebs@webrtc.org <aluebs@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add experimental noise suppression flag to audioproc test
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5608 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 6a94734d4d2f6b83072c18a558b7e1cdc8e6aaae |
|
16-Jan-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds back set_sample_rate_hz() when Init is called in recordings.
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 60730cfe3ce80e4023cd678373456cb703f000a4 |
|
07-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the requirement to call set_sample_rate_hz and friends.
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 22858d4785f245c5a61941c328ffeef6075efb8c |
|
23-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add an extended filter option to audioproc.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| ca764ab22de515f5be5b87cc127e83db32e40208 |
|
07-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a parameter to audioproc for overriding the delay.
Rename the parameter for adding to the input delay to "add_delay".
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2345007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4939 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| f3930e941c15da48c037c62cdb1eebbcbf89c9c7 |
|
19-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Small refactoring of AudioProcessing use in channel.cc.
- Apply consistent naming.
- Use a scoped_ptr for rx_audioproc_.
- Remove now unnecessary AudioProcessing::Destroy().
R=bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4784 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 8c34ceeef176fc0d7850cd9c8973db6d360d86c3 |
|
28-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD
BUG=
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1571004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4119 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 7fad4b8c9f1e9a6e3de9962fb74d4953b4f1bb03 |
|
28-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in audio_processing/
BUG=1662
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| dff69c56b06418a1267a280a9ea419502741ba05 |
|
01-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add AEC suppression level option to audioproc.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/1368007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3927 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 1acb3b33bc3a4225f185acd4e905d9c089b2a5b9 |
|
26-Apr-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add comfort noise disabling and routing mode selection to audioproc.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/1358004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3907 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| b7192b82476d00384fdc153e6a09a6ac53cef67b |
|
10-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in audio_processing/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 6be1e934ad48ccdac734b5cbd50528235ec93816 |
|
01-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly error check calls to AudioProcessing.
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.
BUG=chromium:178040
Review URL: https://webrtc-codereview.appspot.com/1147004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 00ab7cf4fd83b9031325b8e67de606becf4ad920 |
|
11-Feb-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix perf output for audioproc and iSAC fixed-point tests
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1093007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| bafdae3cfccdc962f8c23d7ee25bb63281ef79fa |
|
12-Jan-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix simulated analog gain in audioproc.
* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.
BUG=1260
Review URL: https://webrtc-codereview.appspot.com/1027007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 10abe25f6dcf2d6a3ff579ecc3101d97bad8f605 |
|
17-Dec-2012 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make audioproc output files be written to output dir by default.
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat
TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1003005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 0e739508e08d8f6b254de7e3e9bddd3f8a0e6851 |
|
07-Dec-2012 |
kma@webrtc.org <kma@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added buildbot benchmarking in iSAC and APM into Android platform build.
Review URL: https://webrtc-codereview.appspot.com/964022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| b43502e3889e92e0ed970bfcade0a3555186d754 |
|
27-Nov-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots.
Review URL: https://webrtc-codereview.appspot.com/929022
TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3172 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 4cd8f1f182aaee2f8821c6a3158bd00536d1b732 |
|
26-Nov-2012 |
kma@webrtc.org <kma@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added performance benchmarking in APM and iSAC-fix for Buildbots.
Review URL: https://webrtc-codereview.appspot.com/929022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3170 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 8186534111bf74a8689f63b40dd1f40872bab14d |
|
27-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Only reinitialize AudioProcessing when needed.
This takes away the burden from the user, resulting in cleaner code.
Review URL: https://webrtc-codereview.appspot.com/941005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3010 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 534e495df0bdf2890b10c7ff65e59d96291e1981 |
|
22-Oct-2012 |
leozwang@webrtc.org <leozwang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Qickly fixed android platform build breakage
TBR=ajm
Review URL: https://webrtc-codereview.appspot.com/920006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2966 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/test/process_test.cc
|