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/g711/g711_interface.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/g711/g711_interface.c
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1c6239a3b622fd886d1a2d78cb716b4745446a51 |
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09-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
G711: Make input arrays const and use uint8_t[] for byte arrays BUG=909 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39809004 Cr-Commit-Position: refs/heads/master@{#8294} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8294 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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c78cf97ecb9d8627074b3d64095e5c6cad7da8bb |
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04-Nov-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove the useless dummy state parameter to WebRtcG711_* R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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262e676a08fc29ee6c414f5858d68697be983515 |
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04-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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1b8b4c4959c5a1cf08af527e28eef86940d73880 |
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03-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7041 " Audio codecs to include webrtc/typedefs.h" Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio R=turaj@webrtc.org TBR=andresp@webrtc.org BUG=N/A Review URL: https://webrtc-codereview.appspot.com/19219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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9730d3aae91799334dea86a0439f86fa7c4ab2a5 |
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03-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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621df678c8690f36875b0b34d45393df58662172 |
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22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. Mostly to remove a long-standing TODO... TESTED=trybots R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2369005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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ae4e2b352b4d17c6184687949778704ea60d1da6 |
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21-Mar-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word -> stdint in audio_coding/g711/ BUG= Review URL: https://webrtc-codereview.appspot.com/1223004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3699 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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5ac387c4d1434fc086f3f159bdde068a5994feed |
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19-Nov-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Allow NetEQ to use real packet durations. This is a copy of http://review.webrtc.org/864014/ This adds a FuncDurationEst to each codec instance which estimates the duration of a packet given the packet contents and the duration of the previous packet. By default, this simply returns the duration of the previous packet (which is what is currently assumed to be the duration of all future packets). This patch also provides an initial implementation of this function for G.711 which returns the actual number of samples in the packet. BUG=issue1015 Review URL: https://webrtc-codereview.appspot.com/935016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/g711_interface.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/g711/g711_interface.c
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