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/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
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2224294c52fc0bb95eedb8187a62bedf1b5eb853 |
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03-Jul-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
iSAC: Functions for importing and exporting bandwidth est. info They make it possible to send bandwidth estimation info from decoder to encoder even if they are separate objects (which we want them to be because multithreading). R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1208923002. Cr-Commit-Position: refs/heads/master@{#9535}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
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728d9037c016c01295177fa700fc7927f0bb80bb |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects. This includes changes like: * Attempt to break lines at better positions * Use "override" in more places, don't use "virtual" with it * Use {} where the body is more than one line * Make declaration and definition arg names match * Eliminate unused code * EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT) * Correct #include order * Use anonymous namespaces in preference to "static" for file-scoping * Eliminate unnecessary casts * Update reference code in comments of ARM assembly sources to match actual current C code * Fix indenting to be more style-guide compliant * Use arraysize() in more places * Use bool instead of int for "boolean" values (0/1) * Shorten and simplify code * Spaces around operators * 80 column limit * Use const more consistently * Space goes after '*' in type name, not before * Remove unnecessary return values * Use "(var == const)", not "(const == var)" * Spelling * Prefer true, typed constants to "enum hack" constants * Avoid "virtual" on non-overridden functions * ASSERT(x == y) -> ASSERT_EQ(y, x) BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1172163004 Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
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0946a56023d821e0deca04029bb016ae1f23aa82 |
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09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t etc. in audio_coding/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1271006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
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