| 4d291f7d5e088e3e9f4680dabbe431e62b827f64 |
|
17-Nov-2015 |
peah <peah@webrtc.org> |
Applied the render queueing to the agc.
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1416583003
Cr-Commit-Position: refs/heads/master@{#10667}
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| 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/agc/legacy/analog_agc.c
|
| 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/agc/legacy/analog_agc.c
|
| f6a99e63b6e18d3b3db25e0059a4979743046f31 |
|
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/agc/legacy/analog_agc.c
|
| 1afbdc7555b4bcf9da3bc12ed60e32a0ac0dd297 |
|
10-Mar-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Refactor audio_processing/agc: Removes usage of macro WEBRTC_SPL_MUL_16_16_RSFT
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
BUG=3348,3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47449004
Cr-Commit-Position: refs/heads/master@{#8664}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8664 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| 5e5b32706aa1ac3790feac97e7dc2557845a2a71 |
|
08-Jan-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8023 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| e468bc9e604213054e5fc73431ee127ebe0211a8 |
|
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/agc/legacy/analog_agc.c
|
| cf6d0b64ef8f1d4025f28fd55723be737155ceac |
|
16-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Add 48kHz support to AGC
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/
BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|
| b395a5ea6535338d7486d91128d07bdb95c1c9c4 |
|
16-Dec-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.
BUG=
R=andrew@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
|