| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 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/resampler/sinc_resampler.cc
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| b297c5a01f88219da26cffe433804963d1b70f0f |
|
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/common_audio/resampler/sinc_resampler.cc
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| 728d9037c016c01295177fa700fc7927f0bb80bb |
|
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/resampler/sinc_resampler.cc
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| cb7f8ce2df7564546936d3041a96ccc86a90f988 |
|
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/resampler/sinc_resampler.cc
|
| 073dd7b423157020846872ca2398cc86d3ca9508 |
|
11-Feb-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
WebRtc_GetCPUFeaturesARM is only available on android
R=andrew@webrtc.org, jridges@masque.com, zhongwei.yao@arm.com
Review URL: https://webrtc-codereview.appspot.com/35119004
Patch from Mostyn Bramley-Moore <mostynb@opera.com>.
Cr-Commit-Position: refs/heads/master@{#8336}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8336 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
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| 2ebfac5649a5e48fbbc501b42a4336ff979c03e6 |
|
14-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove COMPILE_ASSERT and use static_assert everywhere
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 3df38b442f6ba29722049b4c4d7121053003a1f8 |
|
13-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Unify the two copies of compile_assert.h
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 3cbd6c26c861a63f8f6164b7142d532a599f00e5 |
|
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/resampler/sinc_resampler.cc
|
| f2aafe4355c4b7ecbd122798f08a5c5ec5d2693a |
|
29-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added include of assert.h for files calling assert but missing the include.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 00073aafa8fde49357181214231cb20e1d763df4 |
|
27-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Clean up CPU detection defines in SincResampler a little.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5615 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 2038920a2b2a36073ddc9739b5a5ab9c6443af2c |
|
26-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness.
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5613 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 12dc1a38ca54a000e4fecfbc6d41138b895c9ca5 |
|
05-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| b86fbaf1d41db539205ec671ff399a3a3aa50734 |
|
26-Jul-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Downstream latest Chromium SincResampler changes.
Replace the BlockSize() workaround we were using previously to support
the push wrapper with the upstream request_frames interface. This
requires a bit of a trick to ensure we don't add more delay than
necessary. On the first pass we use a dummy Resample() call in order to
prime the buffer such that all later calls only require a single input
request through Run().
Notably, this brings in an optimized loop condition, improving
performance by ~2% - 3% on tested platforms and avoids a 20% performance
hit with clang. This addresses issue2041.
Only negligible changes to the PushSincResamplerTest SNR thresholds, due
to a fractional sample adjustment in output delay.
This still retains the per-instance CPU detection, as webrtc lacks a
LazyInstance helper for static initialization.
Ideally, we would adopt SetRatio() in PushSincResampler's
InitializeIfNeeded() for on-the-fly changes, but this will require a way
to update request_frames.
The diff against Chromium upstream is available here:
https://codereview.chromium.org/19470003
BUG=2041
TESTED=unit tests, voe_cmd_test in loopback running through all codecs
with 44.1 kHz and 48 kHz device formats using a stereo mic.
R=dalecurtis@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1838004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4406 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| c6a3755adabc5bf73a8f0cf40c4434fa353e5e33 |
|
08-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update SincResampler with the latest Chromium code.
* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.
TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1438004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 8fc05feed43c702eb84fc26b36aecce33622b06b |
|
26-Apr-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a push-based wrapper around SincResampler.
Includes a unittest to ensure we meet the same quality thresholds as SincResampler (modulo quantization error).
BUG=webrtc:1395
Review URL: https://webrtc-codereview.appspot.com/1323011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3909 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| 076fc12539185da66aee15c2d8fa3d3a094f57b3 |
|
15-Feb-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Modify SincResampler to build in webrtc.
This is the first in a series of CLs to bring arbitrary resampling to webrtc.
* Replace Chromium-specific helpers with their respective webrtc versions.
* Add a second constructor to permit runtime selection of block_size.
* Add stringize_macros to system_wrappers.
BUG=webrtc:1395
TESTED=unit tests
Review URL: https://webrtc-codereview.appspot.com/1097012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|
| a8ef811fe52add24f90973d19a31347226456abc |
|
14-Feb-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Import SincResampler from Chromium.
Committing the originals to make further reviews cleaner.
TBR=bjornv
BUG=webrtc:1395
Review URL: https://webrtc-codereview.appspot.com/1096010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3508 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinc_resampler.cc
|