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/blocker.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/blocker.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/blocker.cc
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d40af69278f867965fd10313f9c99f90266a9469 |
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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
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e534086492e92c45d74b176f3c8be4addb69713f |
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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
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041035b390e25b1d388d73ef52c9e68a550a982f |
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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
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c0da63c7075c3db891b25d193114f7f4f6c7f86a |
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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
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6f10ae25eab4169b44eb1f462ec9d01f5ded8a67 |
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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
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325cff01b42064313b221f2c8819ce01218f5ca4 |
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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
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