15 Commits

Author SHA1 Message Date
kwiberg
c2b785df5d Replace scoped_ptr with unique_ptr in webrtc/common_audio/
(This is a re-land---without the real_fourier.h changes---of 11716, which was reverted in 11726.)

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1731153002

Cr-Commit-Position: refs/heads/master@{#11742}
2016-02-24 13:22:40 +00:00
kjellander
e80f9d0218 Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (patchset #4 id:60001 of https://codereview.webrtc.org/1712513002/ )
Reason for revert:
Breaks downstream compilation using webrtc/common_audio/real_fourier.h. Let's chat tomorrow on how to coordinate a re-land.

Original issue's description:
> Replace scoped_ptr with unique_ptr in webrtc/common_audio/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/79d7a499c0c3e1de8f5ad1138236f0386701053f
> Cr-Commit-Position: refs/heads/master@{#11716}

TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1726043002

Cr-Commit-Position: refs/heads/master@{#11726}
2016-02-23 21:33:39 +00:00
kwiberg
79d7a499c0 Replace scoped_ptr with unique_ptr in webrtc/common_audio/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1712513002

Cr-Commit-Position: refs/heads/master@{#11716}
2016-02-23 09:26:52 +00:00
Peter Kasting
6955870806 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}
2016-01-13 00:26:55 +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
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
wtc@chromium.org
4553941d32 Document the 'int' return value of Resampler methods.
Remove an obsolete TODO comment.

R=andrew@webrtc.org
BUG=none

Review URL: https://webrtc-codereview.appspot.com/48589004

Cr-Commit-Position: refs/heads/master@{#8814}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8814 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 23:28:39 +00:00
kwiberg@webrtc.org
00b8f6b364 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
2015-02-26 14:43:50 +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
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path.
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.

BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 19:00:29 +00:00
pbos@webrtc.org
aa30bb7ef5 Include files from webrtc/.. paths in common_audio/
BUG=1662
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1535005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 09:49:58 +00:00
andrew@webrtc.org
50b2efef6e Add a wrapper around PushSincResampler and the old Resampler.
The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-29 17:27:29 +00:00
pbos@webrtc.org
b09130763b WebRtc_Word32 -> int32_t in common_audio/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3803 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 16:40:28 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00