| 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/noise_suppression_impl.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/noise_suppression_impl.cc
|
| 29e2f9385b7e0dde4af7317af5cd0ce6adf1ee9d |
|
16-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Fix NoiseSuppression initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1507683006/.
BUG=webrtc:5298
Review URL: https://codereview.webrtc.org/1523323002
Cr-Commit-Position: refs/heads/master@{#11043}
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 5e465c33cac54ed5265f18413f7afc44aae2dfca |
|
08-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Make NoiseSuppression not a processing component (bit exact).
BUG=webrtc:5298
patch from issue 1490333004 at patchset 1 (http://crrev.com/1490333004#ps1)
Review URL: https://codereview.webrtc.org/1507683006
Cr-Commit-Position: refs/heads/master@{#10944}
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| df3efa8c079294857a8b8e0a02634d06a6d6b6d6 |
|
28-Nov-2015 |
peah <peah@webrtc.org> |
Introduced the new locking scheme
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1424663003
Cr-Commit-Position: refs/heads/master@{#10836}
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 9b72af94cd61782ada88f777b07854daf9657bb2 |
|
11-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1440523002 .
Cr-Commit-Position: refs/heads/master@{#10608}
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.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/noise_suppression_impl.cc
|
| 9345e86551a0e59e77dfee6eed3e2fe2a833b269 |
|
10-Jun-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
audio_processing: Create now returns a pointer to the object
Affects
* NS
* AGC
* AEC
BUG=441
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1175903002.
Cr-Commit-Position: refs/heads/master@{#9411}
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.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/noise_suppression_impl.cc
|
| c5ebbd98f5996db0defbbfc14f5ca41e620bd7e4 |
|
10-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Support 48kHz in Noise Suppression
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
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| a7384a1126cda7ce726f73b023bad997627fc138 |
|
03-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Simplify audio_buffer APIs
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 5088377d7076cc223992b58f57cc051ceec600e0 |
|
26-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Revert 7297 "Remove the different block lengths in ns_core"
> Remove the different block lengths in ns_core
>
> This CL has bit-exact output.
>
> What it does:
> * Remove the blockLen10Ms, as it is hardcoded to be equal to blockLen.
> * This makes outLen to be always zero, so it can be removed too.
> * It also avoids the need to have an outBuf, because it is not used, so it is also removed
> * Replaced blockLen10Ms by blockLen everywhere, since they were hardcoded to be equal.
> * We don't need to check if outLen is zero, because it always is, so it was removed.
> * Of course, the outBuf needs no initial set or copying around, because it is not used.
>
> BUG=webrtc:3811
> R=bjornv@webrtc.org, kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/30539004
TBR=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7306 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
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| 5f3965783ba70e04dc79efa791145cb99e52b961 |
|
25-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Remove the different block lengths in ns_core
This CL has bit-exact output.
What it does:
* Remove the blockLen10Ms, as it is hardcoded to be equal to blockLen.
* This makes outLen to be always zero, so it can be removed too.
* It also avoids the need to have an outBuf, because it is not used, so it is also removed
* Replaced blockLen10Ms by blockLen everywhere, since they were hardcoded to be equal.
* We don't need to check if outLen is zero, because it always is, so it was removed.
* Of course, the outBuf needs no initial set or copying around, because it is not used.
BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7297 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| fda2c2e810a815d98fe8b03e8c6687d14227b3ff |
|
18-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Add Analyze API to NS
This adds an empty API.
In a next CL I will separate the noise estimation from the Process API and fill this function.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7218 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 12cd4437520588623d3e9840c0fd2c2ace9aae6b |
|
10-Jun-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Noise suppression: Change signature to work on floats instead of ints
Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.
(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)
BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.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/noise_suppression_impl.cc
|
| 5964fe0f86a4f33831d1f4994dbde1b42c93bd81 |
|
22-Apr-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
audio_processing: DestroyHandle() now returns void
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.
BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 56e4a05053d6addc7dbbe2b4d07271305fdbea75 |
|
27-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ProcessingComponent's dependence on AudioProcessingImpl.
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.
BUG=2894
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.cc
|
| 12dc1a38ca54a000e4fecfbc6d41138b895c9ca5 |
|
05-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/noise_suppression_impl.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/noise_suppression_impl.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/noise_suppression_impl.cc
|