| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| e468bc9e604213054e5fc73431ee127ebe0211a8 |
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18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rename _t struct types in audio_processing.
_t names are reserved in POSIX.
R=bjornv@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| c5ebbd98f5996db0defbbfc14f5ca41e620bd7e4 |
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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/ns/ns_core.h
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| ffeaeed8c1cd41f4bd107218b0f60b40cfb10252 |
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28-Oct-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make NSinst_t* const and rename to self in ns_core
This is only to make the code more readable and maintainable.
It generates a bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7550 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| 28b54671cbb066b9b82301a015441491f6c95318 |
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28-Oct-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make all comments whole sentences in ns_core
This is done to make the code more readable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7547 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| b6af4283ca11a639f54d3475bf2a654097f8d287 |
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13-Oct-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Adjust speech probability in NS when echo
The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.
BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| 384d05f362fc0b7ebbb291aaf7ab1f91c1aec56d |
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26-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Remove the different block lengths in ns_core
Relanding the CL: https://webrtc-codereview.appspot.com/30539004/
It had to be reverted because some development code was uploaded by mistake.
TBR=bjornv@webrtc.org
BUG=webrtc:3811
Review URL: https://webrtc-codereview.appspot.com/28589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7307 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| 5088377d7076cc223992b58f57cc051ceec600e0 |
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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/ns/ns_core.h
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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/ns/ns_core.h
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| 275dac2c1d3e8333a200e34e277d3061ba845d3e |
|
24-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Moved the filter calculation from analyze to process in ns_core
It makes sense to have it there if the analyze and process methods are called in different stages.
Tested over the entire QA set for bit exactness.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7287 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| bdfdc96b22bd3ae19dd4d18e17148c69aa1c7285 |
|
22-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Clang-format ns_core
BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| fbf3bfe1722b0af73dd77b2dcfa888597b3a1085 |
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19-Sep-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Separate between Analyze and Process in NS
Filled the empty analyze API, separating the noise estimation from the process API.
No formatting fixes or extra refactoring has been done, to make the review process easier.
This patch has been tested for bit-exactness over the whole QA set in every aggressiveness.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7243 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/ns/ns_core.h
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| 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/ns/ns_core.h
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| 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/ns/ns_core.h
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| 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/ns/ns_core.h
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| 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/ns/ns_core.h
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| 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/ns/ns_core.h
|