| 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/aec/aec_resampler.h
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| 9345e86551a0e59e77dfee6eed3e2fe2a833b269 |
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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/aec/aec_resampler.h
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| f6a99e63b6e18d3b3db25e0059a4979743046f31 |
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10-Apr-2015 |
Bjorn Volcker <bjornv@chromium.org> |
Refactor audio_processing: Free functions return void
There is no point in returning an error when Free() fails. In fact it can only happen if we have a null pointer as object. There is further no place where the return value is used.
Affected components are
- aec
- aecm
- agc
- ns
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50579004
Cr-Commit-Position: refs/heads/master@{#8966}
/external/webrtc/webrtc/modules/audio_processing/aec/aec_resampler.h
|
| 38214d53db5f19228b38424700db8b90cf8b0234 |
|
03-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/aec/aec_resampler.h
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| 13b2d4659380675871e2a232bd3a0b8dbbc35a7a |
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09-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
clang-format audio_processing/aec/*
TBR=bjornv
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/2373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4944 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/aec/aec_resampler.h
|
| 7fad4b8c9f1e9a6e3de9962fb74d4953b4f1bb03 |
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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/aec/aec_resampler.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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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/aec/aec_resampler.h
|