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
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
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* 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
e2fc13e42f
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
2014-02-14 23:12:34 +00:00
60730cfe3c
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
2014-01-07 17:45:09 +00:00
8bf755d5c5
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
2013-09-18 17:40:46 +00:00
fc8aaf02e1
Formalized Real 16-bit FFT for APM.
...
It also prepares for introducing Real 16-bit FFT Neon code from Openmax to SPL. CL https://webrtc-codereview.appspot.com/1819004/ takes care of that, but this CL is a prerequisite of that one.
Tested audioproc with an offline file. Bit exact.
R=andrew@webrtc.org , rtoy@google.com
Review URL: https://webrtc-codereview.appspot.com/1830004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4390 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-24 17:38:23 +00:00
eed919d95d
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
2013-05-30 16:38:36 +00:00
aa30bb7ef5
Include files from webrtc/.. paths in common_audio/
...
BUG=1662
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1535005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 09:49:58 +00:00
8e3b594831
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
2013-05-14 09:24:49 +00:00
e4b6064f8e
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
2013-04-10 18:06:57 +00:00
b09130763b
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
2013-04-09 16:40:28 +00:00
5140e24037
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
2013-02-21 20:12:21 +00:00
ac46c6dac3
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
2013-01-31 21:06:16 +00:00
0af0d3d3f4
Address a build issue with Android-Clang compiler:
...
error: the value is truncated when put into register, use a modifier to specify the size [-Werror,-Wasm-operand-widths]
__asm __volatile ("ssat %0, #16 , %1" : "=r"(out16) : "r"(value32));
Review URL: https://webrtc-codereview.appspot.com/1029006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3352 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-10 00:46:37 +00:00
55cd78cfc2
Porting ARM optimization from Android to ios.
...
Tested APM and iSAC in Android. Bit-exact with original versions.
Changes include removing or changing some GCC derivatives (e.g. .fnstart, .hword), instruction syntax, etc.
Review URL: https://webrtc-codereview.appspot.com/934009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3124 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-17 00:22:46 +00:00
53a8be20f1
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
2012-11-15 08:34:38 +00:00
ff137edc53
Work around with issue 971 (signal_processing_unittests fails memcheck when compiled with GCC 4.6).
...
Review URL: https://webrtc-codereview.appspot.com/935008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3017 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 01:19:42 +00:00
14b43beb7c
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
2012-10-22 18:19:23 +00:00
