8 Commits

Author SHA1 Message Date
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +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
Alejandro Luebs
cdfe20bfc1 Fix the maximum native sample rate in AudioProcessing
BUG=webrtc:4983
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1338833002 .

Cr-Commit-Position: refs/heads/master@{#10037}
2015-09-23 19:49:21 +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
Peter Kasting
b7e5054414 Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones.  For example:

* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps.  For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika

Review URL: https://codereview.webrtc.org/1168753002

Cr-Commit-Position: refs/heads/master@{#9419}
2015-06-11 19:56:03 +00:00
andrew@webrtc.org
f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00
andrew@webrtc.org
a78a41f985 Move output_mixer_unittest.cc to utility_unittest.cc.
This reflects a move of the tested code in:
https://webrtc-codereview.appspot.com/11019005/

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/11449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 23:09:28 +00:00