| 6955870806624479723addfae6dcf5d13968796c |
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13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/common_audio/audio_converter.h
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
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}
/external/webrtc/webrtc/common_audio/audio_converter.h
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| 3c089d751ede283e21e186885eaf705c3257ccd2 |
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16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic 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.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/webrtc/common_audio/audio_converter.h
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| dce40cf804019a9898b6ab8d8262466b697c56e0 |
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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/common_audio/audio_converter.h
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| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
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26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
|
| 2c29c2eae2500c0651e8c074bb5bed042c3ec9d9 |
|
11-Feb-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
C++ readability review for ajm.
As part of the review, refactored AudioConverter into internal derived
classes, each focused on one type of conversion. A factory method
returns the correct converter (or chain of converters, via
CompositionConverter).
BUG=b/18938079
R=rojer@google.com
Review URL: https://webrtc-codereview.appspot.com/35699004
Cr-Commit-Position: refs/heads/master@{#8322}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8322 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
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| 035e9123e9c7de3b09b37bc8e57907e4af7ce219 |
|
28-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move channel_buffer.{h,cc} to common_audio.
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.
BUG=4185
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35939004
Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
|
| 8789376cd35e055765a72248a8ad444ea2e9438c |
|
28-Nov-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Move ChannelBuffer class to channel_buffer file
No change in functionallity.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
|
| 58049360522e2132e5247ddf7cdd4b818d016a45 |
|
03-Nov-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add format members to AudioConverter for DCHECKing.
And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/25059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
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| aada86b261146336d74ea09acedaf40e5c2f4618 |
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27-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a simple AudioConverter class.
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.
The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.
BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/30779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.h
|