| 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.cc
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| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
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17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
/external/webrtc/webrtc/common_audio/audio_converter.cc
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| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/common_audio/audio_converter.cc
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| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
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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.cc
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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.cc
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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.cc
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| 2c29c2eae2500c0651e8c074bb5bed042c3ec9d9 |
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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.cc
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| d35a5c350617cc9d60ce45201764a99229b7299a |
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10-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make ChannelBuffer aware of frequency bands
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/audio_converter.cc
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| 58049360522e2132e5247ddf7cdd4b818d016a45 |
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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.cc
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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.cc
|