874920f22e
Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash.
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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
e2fc13e42f
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
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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.
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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
a3b7406219
Remove unused unreferenced code in webrtc/
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The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.
BUG=
R=andrew@webrtc.org , henrike@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org , turaj@webrtc.org , wu@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1945004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:47:51 +00:00
fc8aaf02e1
Formalized Real 16-bit FFT for APM.
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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
3d8647f17d
Include files from webrtc/.. paths in signal_processing/.
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BUG=1662
R=andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1784004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4352 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:32:03 +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/
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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/
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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
f9c289bafe
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
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* Add the full license to all third party files.
* Correct some entries in LICENSE_THIRD_PARTY which were missing the full
license.
* Relicense all Chromium-licensed files under WebRTC.
* Remove third_party_mods/, which is now redundant.
R=jan.linden@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1396004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3959 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-05 18:54:10 +00:00
342353780d
Consolidate common_audio into a single target.
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In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org , kma@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 23:43:26 +00:00
e4b6064f8e
Replace legacy G_CONST with const.
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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/
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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
7d6f11302e
Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.
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Review URL: https://webrtc-codereview.appspot.com/1153004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3592 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-01 23:01:14 +00:00
5140e24037
MIPS optimizations for Signal Processing Library patch01
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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).
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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
63e0964039
Fix webrtc compilation errors for Chrome Win64
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Mostly disabling warnings in the gyp files.
BUG=1348
BUG=http://crbug.com/166496
BUG=http://crbug.com/167187
Review URL: https://webrtc-codereview.appspot.com/1063011
Patch from Justin Schuh <jschuh@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3423 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 06:45:22 +00:00
a3c82bf667
Remove '<(library)' in gyp files.
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This will remove all usage of '<(library)' in all webrtc gyp files.
Review URL: https://webrtc-codereview.appspot.com/1049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 23:42:21 +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.
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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
f3adba499e
Add Android include path so that header files can follow google style
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BUG=1011
TEST=bot
Review URL: https://webrtc-codereview.appspot.com/930018
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3107 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-15 18:17:40 +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).
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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/
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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