91d6edef35
Add RTC_ prefix to (D)CHECKs and related macros.
...
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
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
ecf6b81644
Pull the Voice Activity Detector out from the AGC
...
This change generates bit-exact values when running through audioproc_f than before.
This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/
And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/
TBR=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1212543002
Cr-Commit-Position: refs/heads/master@{#9505}
2015-06-25 19:28:55 +00:00
51c7cbb86a
Revert "Pull the Voice Activity Detector out from the AGC"
...
This reverts commit 518c683f3e413523a458a94b533274bd7f29992d.
Breaks Linux-Asan bot
https://uberchromegw.corp.google.com/i/client.webrtc/builders/Linux%20Asan/builds/4348/steps/libjingle_peerconnection_unittest/logs/stdio
BUG=
TBR=aluebs@webrtc.org
Review URL: https://codereview.webrtc.org/1208793002 .
Cr-Commit-Position: refs/heads/master@{#9503}
2015-06-25 06:46:14 +00:00
518c683f3e
Pull the Voice Activity Detector out from the AGC
...
This change generates bit-exact values when running through audioproc_f than before.
This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/
And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/
TBR=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1211563003
Cr-Commit-Position: refs/heads/master@{#9502}
2015-06-25 01:46:03 +00:00
f260fc2136
Revert "Pull the Voice Activity Detector out from the AGC"
...
This reverts commit 34be126c1b3ee60ecdb86b1de41a0648347450b2.
It breaks Chromium ASAN.
TBR=niklas.enbom@webrtc.org
Review URL: https://codereview.webrtc.org/1192863006 .
Cr-Commit-Position: refs/heads/master@{#9472}
2015-06-19 18:24:01 +00:00
34be126c1b
Pull the Voice Activity Detector out from the AGC
...
This change generates bit-exact values when running through audioproc_f than before.
R=andrew@webrtc.org , bloch@google.com
Review URL: https://codereview.webrtc.org/1181933002 .
Cr-Commit-Position: refs/heads/master@{#9465}
2015-06-18 19:34:00 +00:00
2c9c83d7ec
Remove non-functional asynchronous resampling mode.
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A few other cleanups, most notably using a sane parameter to specify the
number of channels.
BUG=chromium:469814
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46729004
Cr-Commit-Position: refs/heads/master@{#8894}
2015-03-30 17:08:28 +00:00
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
788acd17ad
Merge audio_processing changes.
...
R=aluebs@webrtc.org , bjornv@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/32769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
