1380e266ff48be9718ce0867cfd65058cb09c5fc |
|
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/common_audio/signal_processing/include/signal_processing_library.h
|
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/signal_processing/include/signal_processing_library.h
|
72cfd6c468663a95962d2677c94d91d97b71f980 |
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23-Jun-2015 |
pkasting <pkasting@chromium.org> |
Reland remaining bits of "Upconvert various types to int." Most of commit cb180976dd0e9672cde4523d87b5f4857478b5e9 (which reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24) was already re-landed. This relands the rest, including modifications by kwiberg to hopefully avoid regressing performance. In a subsequent change I will see if removing the int16_t cast in this modified version still causes perf problems. BUG=499241 TEST=none TBR=andrew Review URL: https://codereview.webrtc.org/1181693005 Cr-Commit-Position: refs/heads/master@{#9487}
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
bc440d56513eeb15e1e31b5c5232e215556ffbbe |
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12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Revert "Reland "Upconvert various types to int.", common_audio portion." This reverts commit 15b58eea282b03b6347c64714079691f55e6097f. BUG=499241 TBR=andrew Review URL: https://codereview.webrtc.org/1182683003 Cr-Commit-Position: refs/heads/master@{#9426}
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
15b58eea282b03b6347c64714079691f55e6097f |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", common_audio portion. This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the files in webrtc/common_audio/ 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=andrew Review URL: https://codereview.webrtc.org/1184613003 Cr-Commit-Position: refs/heads/master@{#9425}
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
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/common_audio/signal_processing/include/signal_processing_library.h
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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/common_audio/signal_processing/include/signal_processing_library.h
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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/common_audio/signal_processing/include/signal_processing_library.h
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cb7f8ce2df7564546936d3041a96ccc86a90f988 |
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20-May-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Clear ARM NEON flag Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one WEBRTC_HAS_NEON. Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON. Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu. BUG=4002 R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980 Review URL: https://webrtc-codereview.appspot.com/49309004 Cr-Commit-Position: refs/heads/master@{#9228}
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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1ccd8b4281f8d3ebbc51e70a96ac186e787a7114 |
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25-Mar-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT The macro is defined as #define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \ (WEBRTC_SPL_MUL_16_16(a, b) >> (c)) where the latter macro is in C defined as #define WEBRTC_SPL_MUL_16_16(a, b) \ ((int32_t) (((int16_t)(a)) * ((int16_t)(b)))) (For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h) The replacement consists of - avoiding casts to int16_t if inputs already are int16_t - adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t) - minor cleanups like remove of unnecessary parentheses and style changes BUG=3348, 3353 TESTED=locally on Linux for both fixed and floating point and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49499004 Cr-Commit-Position: refs/heads/master@{#8853}
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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0933d01d094648f6c133e9d5d71e27da4e013e54 |
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05-Mar-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Enabling common_audio building with NEON on ARM64 Passed building common_audio_neon and common_audio_unittests both on Android ARMv7 and Android ARM64. Pass common_audio_unittests tests both on Android ARMv7 and Android ARM64. BUG=4002 R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org Change-Id: I8e0722f356db8cca6fc8232f00ae1e898a086f5a Review URL: https://webrtc-codereview.appspot.com/40629004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. Cr-Commit-Position: refs/heads/master@{#8620} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8620 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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88a429823448ff4928bbdda6e2d4ce2576474678 |
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12-Jan-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin() In addition, expanded the unit test to verify both unstable and stable filters. BUG=3353, 1132 TESTED=locally on Mac and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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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/common_audio/signal_processing/include/signal_processing_library.h
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d730b288c8873d6459250327be02c1895c86e652 |
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07-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon This function isn't used anymore. The file and header are also removed. BUG=4002,3273 R=andrew@webrtc.org Change-Id: I4b65dec57e6adc2ac2253031501f3b6de6937fac Review URL: https://webrtc-codereview.appspot.com/35519004 Patch from Yang Zhang <yang.zhang@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8019 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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f567095f620ac9961b2d5b006b239f3a08fe576f |
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29-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 Replaces the trivial macro WEBRTC_SPL_RSHIFT_W32 with >> at various places in common_audio and removes it. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7558 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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67ca26e08798374ed34a2635fe5d2a09303348ba |
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28-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t. Affected components: * isac/fix BUG=3348,3353 TESTED=locally on linux and trybots R=henrik.lundin@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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fab543911287c699fce2deef4aae691e982ae8f0 |
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15-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed version API from signal_processing The Signal Processing version API is not used anymore. BUG=3353 R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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bc1a4578e0a8dd98182efb166f6f03d09ca9091d |
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13-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing. Affected components: * vad * aecm BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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3ea35fdb1b3bc50e98b0fee03d274dce76fbdd55 |
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09-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16 The macro was a trivial << operation and where used has been replaced by <<. Affected components are * ilbc * isacfix BUG=3348,3353 TESTED=locally on linux and trybots R=henrik.lundin@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22919005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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7c15510f389b00fea03e8512cf1a09d0a344b8e9 |
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29-Sep-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary. Affects both fixed and float point versions of iSAC BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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3cbd6c26c861a63f8f6164b7142d532a599f00e5 |
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04-Sep-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. This changes some method signatures to better reflect how callers are actually using them. This also has the tendency to make signatures more consistent about e.g. using int (instead of int16_t) for lengths of things like vectors, and using int16_t (instead of int) for e.g. counts of bits in a value. This also removes a couple of functions that were only called in unittests. BUG=3353,chromium:81439 TEST=none R=andrew@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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df9fef6638b4b5ef7a6c9a3eb2b20e26aefc9746 |
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28-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio: Removed macro WEBRTC_SPL_DIV The macro has no built-in divide by zero check. The only thing that is done is casting to int32_t. In addition a bug was discovered where it was supposed to do a division with rounding, but instead did a division with truncation + addition by 2. This is corrected in this CL. BUG=3348,3353 TESTED=locally on Linux R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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4f71e22bf97903ba52bed537a3f614089ce1fb93 |
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26-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use. This CL replaces the macro with '/' at place. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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926707b167e9bfb876c7c903dcefc41c84e5fedb |
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25-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Refactoring common_audio: Replace trivial multiplication macro This multiplication macro literally use the '*' operator, so there is no need for it. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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d32c4389ac4642e2e8c9b5ec3096f57290a346f6 |
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25-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Re-landing r6961 common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. BUG=3348,3353 TESTED=locally on linux TBR=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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4a616be12b9a941076892af89625f847513999dc |
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25-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..." > common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 > > This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. > > BUG=3348,3353 > TESTED=locally on linux and trybots > R=kwiberg@webrtc.org, tina.legrand@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/16359004 TBR=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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4f01017e2d23c97e64170569f631899301319575 |
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25-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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6e71d17bc98d3b179f55fd4fd42dc5c53b7787dc |
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25-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Refactoring common_audio/signal_processing: Replaces trivial macros The macros WEBRTC_SPL_ADD_SAT_W16 and WEBRTC_SPL_ADD_SAT_W32 make direct use of the corresponding functions WebRtcSpl_AddSatW16() and WebRtcSpl_AddSatW32(). This CL replaces these macros in the code. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6960 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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52275341d82ef7839d902971242d7ccd6f972043 |
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20-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16 Yet another macro that utilizes a function directly. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18159004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6935 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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b5ab52d010f210bd868230ed7d07d9317eb76e6f |
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18-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and WEBRTC_SPL_SET_BYTE These two macros are not used anywhere in webrtc. Previously used in old neteq (I think). BUG=3348,3353 TESTED=manually on linux and trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6916 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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1e3ef4b999ada52fc4e421d1d06b73068e855413 |
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15-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16 Macros should in general be avoided. WEBRTC_SPL_UMUL_32_16_RSFT16 is only used in iSAC fixed point as part of multiplying with LSB and MSB. A better approach is to have one function for that complete operation in iSAC. This CL removes the macro and replace the operation locally. BUG=3148, 3353 TESTED=locally on Linux and trybots R=tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6907 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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8434dbe28417cce44d8f9181581092b67b0e1e56 |
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14-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32 This macro is literally using the function WebRtcSpl_SubSatW32(), hence there is no need for a macro. BUG=3348, 3353 TESTED=locally on Linux and trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6899 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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0a3cbb3906c588d089a56c157f6e82b7a4d404d4 |
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12-Aug-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32 The macro is only used at four places in iSAC fixed point and the macro have been replaced at those places. In addition, it is used in a unit test, but throws a warning treated as error (issue3674). The macro has both MIPS and armv7 optimizations. Removing them impacts only MIPS platforms without DSP ASE. This may cause a very small increase in complexity when using iSAC fix. The armv7 optimizations are not used anywhere, since specific ones are used inline in iSAC fix. BUG=3348,3353,3674 TESTED=locally and trybots R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6871 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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c0ba4392f18f083839673c97f3f63015a72a8bb5 |
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03-Jul-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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721f970cba4a9987d1e42e267e26df8414deae99 |
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16-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift. Related CL: https://webrtc-codereview.appspot.com/16669004 BUG=3348,3353 TESTED=trybots and manually on linux R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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18026abd823827b4bf818ba2387b87950ce42fd9 |
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11-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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782978cfcb6fa60f879f2b80088b5c5d69ea27c5 |
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11-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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edbe886a0ba31422b130feee430e82a4051c2d3d |
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05-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation. BUG=3348,3353 TESTED=trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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a3b5673879284f16a2f9f56bc14429eec4f20a0c |
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16-May-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed. BUG=3348 TESTED=trybots, manual unittests R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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1b21a579028998e717c35e320cd1d6258b2b650d |
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16-May-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 Macro was only mapping a function used in one place. BUG=3348 TESTED=trybots, unittests R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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d83d607271bf38dc0f18b6cdc5db8fe6c9d35a91 |
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16-May-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED * Moved the macro to randomization_functions and made it static const. * Made WebRtc_IncreaseSeed() static, since it is not used outside this function. * Style guide changes. BUG=3348,3353 TESTED=trybots, common_audio_unittests, modules_unittests, modules_tests R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6179 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
1aae6bf73554bfac32bdb87b6a07bb12d05d8e82 |
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15-May-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
common_audio: Removes unused macros * WEBRTC_SPL_MUL_32_32_RSFT32BI * WEBRTC_SPL_IS_NEG BUG=3348 TESTED=trybots, common_audio_unittests R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6169 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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e2fc13e42f3881d2833af627a23ac77e938515ec |
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15-Feb-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring common_audio/signal_processing: Removed two macros used by isac only. Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation. Further, the includes were updated with full paths and put in alphabetical order. BUG=N/A TESTED=trybots,module_tests,module_unittests R=turaj@webrtc.org, turajs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
60730cfe3ce80e4023cd678373456cb703f000a4 |
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07-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the requirement to call set_sample_rate_hz and friends. Instead have ProcessStream transparently handle changes to the stream audio parameters (sample rate and channels). This removes two locks per 10 ms ProcessStream call taken by VoiceEngine (four total with the audio level indicator.) Also, prepare future improvements by having the splitting filter take a length parameter. This will allow it to work at different sample rates. Remove the useless splitting_filter wrapper. TESTED=voe_cmd_test with audio processing enabled and switching between codecs; unit tests. R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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8bf755d5c5dd03b3335b22cd687749e140b4aa40 |
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18-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file. R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1791004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@4779 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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eed919d95d6f94a8cb001effc9ddc8b94963164a |
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30-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
MIPS optimizations for the following functions: WebRtcSpl_ComplexBitReverse, WebRtcSpl_ComplexFFT, WebRtcSpl_ComplexIFFT, WebRtcSpl_DownsampleFast and WebRtcSpl_FilterARFastQ12. Also, moved the common table used in complex_fft functions to a separate header file (webrtc/common_audio/signal_processing/include/complex_fft_tables.h). R=andrew@webrtc.org, kma@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1126004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@4141 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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8e3b5948319b3aaf47e3412ce370cb0adefdc7dc |
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14-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove const for plain data types in common_audio/ BUG=1644 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1464005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4019 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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e4b6064f8e6b0b448f070155b1b6c4b6788107df |
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10-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace legacy G_CONST with const. BUG=1608 Review URL: https://webrtc-codereview.appspot.com/1310005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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b09130763b8949023a061f3cfa665a3a6bbac0f0 |
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09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in common_audio/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3803 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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5140e240372d85aaf6c51476b4f1bad92e61bf80 |
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21-Feb-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
MIPS optimizations for Signal Processing Library patch01 Review URL: https://webrtc-codereview.appspot.com/1028004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@3557 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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ac46c6dac3d8de1f5e30841d74839bd5b7285fe2 |
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31-Jan-2013 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replaced relative path to reference from <(webrtc_root). Changed to proper include paths in AECM and NSX. Tested on trybots. BUG=None Review URL: https://webrtc-codereview.appspot.com/1063014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3450 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
|
53a8be20f13eeb84fc2df17abcebf4fcd3b2930d |
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15-Nov-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wraparound distortion in Opus This CL solves the wraparound distortion in Opus. In the Opus decoder-wrapper we are downsampling the signal from 48 kHz to 32 kHz. This is done in two steps, using the following functions from the signal processing library: WebRtcSpl_Resample48khzTo32khz() and WebRtcSpl_VectorBitShiftW32ToW16 The latter does not have a saturation check, and the signal can suffer from wraparound. I've added saturation control to the function. BUG=issue1089 Review URL: https://webrtc-codereview.appspot.com/967004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3103 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/signal_processing/include/signal_processing_library.h
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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/common_audio/signal_processing/include/signal_processing_library.h
|