History log of /external/webrtc/webrtc/common_audio/blocker.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
6955870806624479723addfae6dcf5d13968796c 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/blocker.cc
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/blocker.cc
dce40cf804019a9898b6ab8d8262466b697c56e0 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/blocker.cc
d40af69278f867965fd10313f9c99f90266a9469 28-Jul-2015 andrew <andrew@webrtc.org> Split MoveReadPosition into Forward and Backward versions.

This makes the interface consistent in its use of size_t, and will
reduce casts in callers as they shift to size_t, e.g. this CL:
https://codereview.webrtc.org/1227203003/

Review URL: https://codereview.webrtc.org/1252943007

Cr-Commit-Position: refs/heads/master@{#9646}
/external/webrtc/webrtc/common_audio/blocker.cc
e534086492e92c45d74b176f3c8be4addb69713f 13-Mar-2015 mgraczyk@chromium.org <mgraczyk@chromium.org> Clean up LappedTransform and Blocker.

- Remove unnecessary window member from lapped_transform.
- Add comment indicated that Blocker does not take ownership of
the window passed to its constructor.
- Streamline LappedTransform constructor so members can be const.

Also use a range-based for loop in audio_processing_impl.cc for clarity.

R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41229004

Cr-Commit-Position: refs/heads/master@{#8708}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8708 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/blocker.cc
041035b390e25b1d388d73ef52c9e68a550a982f 26-Jan-2015 andrew@webrtc.org <andrew@webrtc.org> Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.

Integrate it in Blocker to demonstrate use.

TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com

Review URL: https://webrtc-codereview.appspot.com/36799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/blocker.cc
c0da63c7075c3db891b25d193114f7f4f6c7f86a 13-Jan-2015 aluebs@webrtc.org <aluebs@webrtc.org> Optimize minimum delay in blocker

Could not hear any difference when running the beamformer_test, although sample-wise it changes because of the non-linear character of the processing.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8051 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/blocker.cc
6f10ae25eab4169b44eb1f462ec9d01f5ded8a67 17-Dec-2014 aluebs@webrtc.org <aluebs@webrtc.org> Support block_size greater than chunk_size in Blocker

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7934 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/blocker.cc
325cff01b42064313b221f2c8819ce01218f5ca4 01-Oct-2014 andrew@webrtc.org <andrew@webrtc.org> Import LappedTransform and friends.

Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/blocker.cc