15 Commits

Author SHA1 Message Date
kwiberg
9e2be5f292 webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()
Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}
2016-09-14 12:23:29 +00:00
kwiberg
8cf8898973 Fix Chromium clang plugin warnings
BUG=webrtc:163

Review-Url: https://codereview.webrtc.org/2285713002
Cr-Commit-Position: refs/heads/master@{#13943}
2016-08-26 21:50:42 +00:00
peah
bbe423312d Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files
The changes are done in several patches in order to make
the review easier.

NOTRY=True
BUG=webrtc:5949

Review-Url: https://codereview.webrtc.org/2051443002
Cr-Commit-Position: refs/heads/master@{#13068}
2016-06-08 13:42:08 +00:00
Henrik Kjellander
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
henrikg
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
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
aluebs
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
Bjorn Volcker
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
aluebs
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
Alejandro Luebs
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
Alejandro Luebs
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
Andrew MacDonald
2c9c83d7ec Remove non-functional asynchronous resampling mode.
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
kwiberg@webrtc.org
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
kwiberg@webrtc.org
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
pbos@webrtc.org
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