| 4376648df021fd82f25a38694e33678f802d06ea |
|
27-Aug-2015 |
Karl Wiberg <kwiberg@google.com> |
AudioDecoder: Replace Init() with Reset()
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.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_coding/codecs/g722/g722_interface.c
|
| bba780707848ad36066739c60d7b28cd752fb92f |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093003
Cr-Commit-Position: refs/heads/master@{#9424}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| cb180976dd0e9672cde4523d87b5f4857478b5e9 |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Revert "Upconvert various types to int."
This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.
BUG=499241
TBR=hlundin
Review URL: https://codereview.webrtc.org/1179953003
Cr-Commit-Position: refs/heads/master@{#9418}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
|
10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54629004
Cr-Commit-Position: refs/heads/master@{#9405}
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| a1dfbf1e5c9e657adbc32254c169704e5bc7c370 |
|
02-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcG722_Decode: Input array should be const uint8_t[]
BUG=909
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38799004
Cr-Commit-Position: refs/heads/master@{#8224}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8224 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| eb544460e47140d494dddf1217a698a1dcf4dee0 |
|
17-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rename _t struct types in audio_coding.
_t names are reserved in POSIX.
R=henrik.lundin@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 0cd5558f2b9357914873479e7901de6adc44609c |
|
02-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
AudioEncoder subclass for G722
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 262e676a08fc29ee6c414f5858d68697be983515 |
|
04-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Reland rev 7041 with BUILD.gn files.
Original description:
Audio codecs to include webrtc/typedefs.h
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h
CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"
BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 1b8b4c4959c5a1cf08af527e28eef86940d73880 |
|
03-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7041 " Audio codecs to include webrtc/typedefs.h"
Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio
R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/19219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 9730d3aae91799334dea86a0439f86fa7c4ab2a5 |
|
03-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Audio codecs to include webrtc/typedefs.h
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h
CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"
BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 0946a56023d821e0deca04029bb016ae1f23aa82 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t etc. in audio_coding/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1271006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
|
| 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_coding/codecs/g722/g722_interface.c
|