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Author SHA1 Message Date
Peter Kasting
dce40cf804 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}
 2015-08-24 21:52:45 +00:00
kwiberg@webrtc.org
648f5d6dc7 pcm16b: Make input arrays const and use uint8_t[] for byte arrays 
There were both uint8 and uint16 versions of the pcm16b encode and
decode functions; this patch removes the latter.

BUG=909
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34139004

Cr-Commit-Position: refs/heads/master@{#8309}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 09:19:09 +00:00
henrik.lundin@webrtc.org
fcbe36a1d9 Add const qualifier to WebRtcPcm16b_Encode 
BUG=909
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 18:26:49 +00:00
kwiberg@webrtc.org
6de75ca3ed Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:29:24 +00:00
andresp@webrtc.org
262e676a08 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
 2014-09-04 13:28:48 +00:00
henrike@webrtc.org
1b8b4c4959 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
 2014-09-03 19:42:16 +00:00
andresp@webrtc.org
9730d3aae9 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
 2014-09-03 14:37:18 +00:00
bjornv@webrtc.org
d32c4389ac Re-landing r6961 
common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8

This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 11:19:05 +00:00
bjornv@webrtc.org
4a616be12b Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..." 
> common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8
> 
> This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.
> 
> BUG=3348,3353
> TESTED=locally on linux and trybots
> R=kwiberg@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/16359004

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 10:32:22 +00:00
bjornv@webrtc.org
4f01017e2d common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 
This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 10:23:22 +00:00
henrik.lundin@webrtc.org
9b8102cf0e Use a deterministic input in NetEqBgnTest 
This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.

The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)

BUG=3715
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 08:27:44 +00:00
andrew@webrtc.org
621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. 
Mostly to remove a long-standing TODO...

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 10:27:23 +00:00
pbos@webrtc.org
0946a56023 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
 2013-04-09 00:28:06 +00:00
andrew@webrtc.org
14b43beb7c 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
 2012-10-22 18:19:23 +00:00