Commit Graph

300 Commits

Author SHA1 Message Date
047a46f8b4 Remove Android.mk build files.
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
926707b167 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
2014-08-25 11:42:42 +00:00
d32c4389ac 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
2014-08-25 11:19:05 +00:00
4a616be12b 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
2014-08-25 10:32:22 +00:00
4f01017e2d 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
2014-08-25 10:23:22 +00:00
6e71d17bc9 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
2014-08-25 07:44:52 +00:00
584cd8da4b Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)
And in the process, make it dump WAV files instead of raw PCM.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 06:26:04 +00:00
6b2659c660 Refactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_RSFT32BI
The WEBRTC_SPL_MUL_32_32_RSFT32BI macro was removed in r6169, since it was unused. This CL removes the arm and mips optimizations of it.

BUG=3348, 3353
TESTED=locally and trybots
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 06:13:57 +00:00
52275341d8 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
2014-08-20 10:09:34 +00:00
877083c4d4 New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.

This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.

Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.

R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
b5ab52d010 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
2014-08-18 12:01:02 +00:00
1e3ef4b999 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
2014-08-15 05:17:20 +00:00
8434dbe284 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
2014-08-14 07:26:28 +00:00
0a3cbb3906 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
2014-08-12 10:54:50 +00:00
cf8f33a6d6 Removes mismatching signs in signal_processing_unittests
Negative inputs was used in WebRtcSpl_NormU32() causing warnings.

BUG=3674
TESTED=locally and trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12 10:27:21 +00:00
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 21:06:30 +00:00
efb81d8d1f int16<->float conversions: Use size_t for array length argument, not int
size_t is more appropriate for array lengths, since int might
theoretically be too small for a really large array. But more
importantly, if the caller's value is naturally of type size_t and the
function requires an int, VC++ will trigger warning C4267
(http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the
implicit cast might be lossy, forcing the caller to do a manual cast.
Typing the function with size_t in the first place resolves the
problem.

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 08:36:52 +00:00
c0ba4392f1 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
2014-07-03 13:38:53 +00:00
1227ab89a7 GN: Add BUILD.gn files + kjellander to OWNERS
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.

I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.

I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

R=brettw@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
721f970cba 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
2014-06-16 10:30:14 +00:00
a1bfc50a72 Pass GYP DEPTH variable to isolate.
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 09:02:15 +00:00
18026abd82 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
2014-06-11 06:53:20 +00:00
782978cfcb 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
2014-06-11 06:39:03 +00:00
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 05:42:53 +00:00
aafd7a88c5 The correct fix of workaround in r6261.
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c

BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org, glaznev@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:53:51 +00:00
edbe886a0b 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
2014-06-05 08:42:53 +00:00
e6e139159f Android: cleanup gtest_target_type conditions.
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
aca5939dfc common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8
In r6240 gcc was rolled from 4.6 to 4.8 changing the behavior on arm. The output of ComplexFFT differs causing both AECM and NS to perform worse. Looking at issues on gcc it says that there could be a memory shuffling/optimization despite using volatile affecting the output.
Splitting the three instructions in one call into two separate calls makes the compiler take proper actions resulting in correct outputs.

BUG=3370,3395
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6261 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 08:45:04 +00:00
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
be4ab99a53 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
BUG=3370
R=bjornv@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 12:42:01 +00:00
a3b5673879 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
2014-05-16 12:11:20 +00:00
1b21a57902 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
2014-05-16 06:40:31 +00:00
d83d607271 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
2014-05-16 06:38:47 +00:00
1aae6bf735 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
2014-05-15 07:22:53 +00:00
65f933899b Fix constness of AudioBuffer accessors.
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 16:44:13 +00:00
f2aafe4355 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
2014-04-29 17:54:17 +00:00
874920f22e Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash.
The C (unoptimized) version of this function takes less than 1% of NetEq4 and
has been crashing since the NetEq3->4 move.

BUG=3243
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6003 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 20:09:09 +00:00
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>.
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
2a796720f8 common_audio: VADFree() now returns void
* Files in audio_coding are not affected since they never use the return value.
* voice_detection in audio_processing does.
* Updated vad_unittest.cc

BUG=441
TESTED=trybots
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5948 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 04:45:35 +00:00
f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
37ca765650 Add fir_filter to common_audio
It has 3 implementation:
* fir_filter_c with no optimization
* fir_filter_sse which outperforms the C version by a factor of 3x
* fir_filter_neon which outperforms the C version by a factor of 2x

R=andrew@webrtc.org, bjornv@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/9759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5756 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:16:11 +00:00
d32797f853 Add a float interface to PushSincResampler.
Provides a push interface to SincResampler without the int16->float
overhead. This is required to support resampling in the new
AudioProcessing float path.

BUG=2894
TESTED=unit tests
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5673 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 18:51:42 +00:00
17e40641b3 Add a deinterleaved float interface to AudioProcessing.
This is mainly to support the native audio format in Chrome. Although
this implementation just moves the float->int conversion under the hood,
we will transition AudioProcessing towards supporting this format
throughout.

- Add a test which verifies we get identical output with the float and
int interfaces.
- The float and int wrappers are tasked with conversion to the
AudioBuffer format. A new shared Process/Analyze method does most of
the work.
- Add a new field to the debug.proto to hold deinterleaved data.
- Add helpers to audio_utils.cc, and start using numeric_limits.
- Note that there was no performance difference between numeric_limits
and a literal value when measured on Linux using gcc or clang.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 20:58:13 +00:00
00073aafa8 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
2014-02-27 04:12:34 +00:00
2038920a2b 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
2014-02-26 18:14:54 +00:00
d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.

TESTED=try bots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 21:08:36 +00:00
d4d5be8781 Minor improvement in RoundToInt16 implementation.
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5586 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 20:55:21 +00:00