e84e96e8be15fb80a1cf37ba6ef70c01cc94199d |
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12-Jan-2016 |
Henrik Lundin <henrik.lundin@webrtc.org> |
NetEq: Fix a typo in a comment TBR=minyue@webrtc.org NOTRY=true Review URL: https://codereview.webrtc.org/1578223003 . Cr-Commit-Position: refs/heads/master@{#11226}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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1380e266ff48be9718ce0867cfd65058cb09c5fc |
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29-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Convert some more things to size_t. These changes stem from requests by Andrew on https://codereview.webrtc.org/1228823002/ to eliminate some "return -1"s and change to using asserts plus returning size_ts. I then also converted the relevant connected bits. This also cleans up a bunch of style issues, e.g. no spaces around operators. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://codereview.webrtc.org/1305983003 . Cr-Commit-Position: refs/heads/master@{#9813}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.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/modules/audio_coding/neteq/expand.cc
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bef77e234fa53a52b830b5833948711f75ab8bbb |
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18-Aug-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
NetEq: Implement logging of Delayed Packet Outage Events Measures the duration of each packet loss concealment (a.k.a. expand) event that is not followed by a merge operation. Having decoded and played packet m−1, the next expected packet is m. If packet m arrives after some time of packet loss concealment, we have a delayed packet outage event. However, if instead packet n>m arrives, we have a lost packet outage event. In NetEq, the two outage types results in different operations. Both types start with expand operations to generate audio to play while the buffer is empty. When a lost packet outage happens, the expand operation(s) are followed by one merge operation. For delayed packet outages, merge is not done, and the expand operations are immediately followed by normal operations. This change also includes unit tests for the new statistics. BUG=webrtc:4915, chromium:488124 R=minyue@webrtc.org Review URL: https://codereview.webrtc.org/1290113002 . Cr-Commit-Position: refs/heads/master@{#9725}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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b297c5a01f88219da26cffe433804963d1b70f0f |
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23-Jul-2015 |
pkasting <pkasting@chromium.org> |
Miscellaneous changes split from https://codereview.webrtc.org/1230503003 . These are mostly trivial changes and are separated out just to reduce the diff on that change to the minimum possible. Note explanatory comments on patch set 1. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1235643003 Cr-Commit-Position: refs/heads/master@{#9617}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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36b7cc32643bae0379d8102ce05dae82ecc466a1 |
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12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", neteq portion. This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the files in webrtc/modules/audio_coding/neteq/ are relanded. The original commit message is below: Upconvert various types to int. Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t. Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C." This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change. BUG=none TBR=kwiberg Review URL: https://codereview.webrtc.org/1181073002 Cr-Commit-Position: refs/heads/master@{#9427}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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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/neteq/expand.cc
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b7e5054414ff524f9db81dab7917729b8c4c8bcb |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better. This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example: * Change a few type declarations to better match how the majority of code uses those objects. * Eliminate "< 0" check for unsigned values. * Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar. * Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects. * Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t. * Similarly, add casts when passing a larger type to a function taking a smaller one. * Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar. * Use "false" instead of "0" for setting a bool. * Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t. BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=andrew, asapersson, henrika Review URL: https://codereview.webrtc.org/1168753002 Cr-Commit-Position: refs/heads/master@{#9419}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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cb180976dd0e9672cde4523d87b5f4857478b5e9 |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Revert "Upconvert various types to int." This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. BUG=499241 TBR=hlundin Review URL: https://codereview.webrtc.org/1179953003 Cr-Commit-Position: refs/heads/master@{#9418}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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f045e4da43e671ae511aa1d9b6ef2968256a745d |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Prepare to convert various types to size_t. This makes some behaviorally-invariant changes to make certain code that currently only works correctly with signed types work safely regardless of the signedness of the types in question. This is preparation for a future change that will convert a variety of types to size_t. There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants. BUG=none R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=ajm Review URL: https://codereview.webrtc.org/1174813003 Cr-Commit-Position: refs/heads/master@{#9413}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
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10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Upconvert various types to int. Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t. Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C." This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change. BUG=none R=andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/54629004 Cr-Commit-Position: refs/heads/master@{#9405}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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7f6c4d42a2605d1da39af3f957a46cf57b043b84 |
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09-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/modules/audio_coding/neteq Mostly this consists of marking functions with override when applicable, and moving function bodies from .h to .cc files. BUG=163 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44109004 Cr-Commit-Position: refs/heads/master@{#8960}
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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c14e3572c6c1400aa8ccd909893fc996b9b0ef09 |
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12-Jan-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12() BUG=3353, 1133 TESTED=locally on Mac and trybots R=henrik.lundin@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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a5ce7bbe1781b23474ee5acdddea4f0c443220a5 |
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20-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> BUG=3348,3353 TESTED=locally on linux and trybots R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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f6ab6f86e70be38772d2df798e8e0041dc297562 |
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04-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Rename Audio[Multi]Vector.CopyFrom to .CopyTo The name of the copy method was confusing. This change makes the code easier to read where the method is used. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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023f12fb6ef3f7f844a515fe1b97ff2b3753a9f6 |
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13-Aug-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
NetEq background noise generation off by default This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable. With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise. Reference files are updated where needed. BUG=3519 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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ea25784107b9202300a57d838d2c56e158220eef |
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07-Aug-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change how background noise mode in NetEq is set This change prepares for switching default background noise (bgn) mode from on to off. The actual switch will be done later. In this change, the bgn mode is included as a setting in NetEq's config struct. We're also removing the connection between playout modes and bgn modes in ACM. In practice this means that bgn mode will change from off to on for streaming mode, but since the playout modes are not used it does not matter. BUG=3519 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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9c55f0f957534144d2b8a64154f0a479249b34be |
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09-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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1b9df05c8521d1d807b08d7c00eb2f7e5b097fdf |
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28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6257 "Rename neteq4 folder to neteq" > Rename neteq4 folder to neteq > > BUG=2996 > R=turaj@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/12569005 TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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a90f6d67f72359cf63b59480fa87a13aae808c03 |
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28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq BUG=2996 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12569005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/expand.cc
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