admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
7,477 Commits 1 Branch 0 Tags
Commit Graph

167 Commits

Author SHA1 Message Date
kjellander@webrtc.org
035e9123e9 Move channel_buffer.{h,cc} to common_audio. 
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.

BUG=4185
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35939004

Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:57:44 +00:00
andrew@webrtc.org
922cfcd150 Use non-zero data in AudioRingBufferTest. 
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35909004

Cr-Commit-Position: refs/heads/master@{#8176}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 21:59:44 +00:00
andrew@webrtc.org
041035b390 Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. 
Integrate it in Blocker to demonstrate use.

TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com

Review URL: https://webrtc-codereview.appspot.com/36799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 21:23:53 +00:00
kjellander@webrtc.org
d7e34e1086 Make it easier to use external libyuv + cleanup GYP files. 
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 19:17:26 +00:00
bjornv@webrtc.org
d25c034051 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() 
BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:32:47 +00:00
mgraczyk@chromium.org
5a92b78e86 Add beamforming to audioproc_float utility. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 01:28:36 +00:00
andrew@webrtc.org
6b6301588e Move ring_buffer to common_audio. 
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.

R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 00:09:53 +00:00
kwiberg@webrtc.org
2ebfac5649 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
 2015-01-14 10:51:54 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +00:00
aluebs@webrtc.org
c0da63c707 Optimize minimum delay in blocker 
Could not hear any difference when running the beamformer_test, although sample-wise it changes because of the non-linear character of the processing.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 22:28:35 +00:00
kwiberg@webrtc.org
3df38b442f 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
 2015-01-13 11:37:48 +00:00
bjornv@webrtc.org
88a4298234 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
 2015-01-12 05:53:43 +00:00
bjornv@webrtc.org
c14e3572c6 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
 2015-01-12 05:50:52 +00:00
andrew@webrtc.org
d730b288c8 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
 2015-01-07 21:34:23 +00:00
andrew@webrtc.org
84d84471f5 Minor fixes regarding accumulator usage on MIPS platforms. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 17:08:44 +00:00
andrew@webrtc.org
1090a6eccf Remove obsolete target_arch == armv7. 
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 21:36:18 +00:00
andrew@webrtc.org
0ab42bc3f6 Make safe_conversions suitable for rtc_base_approved. 
Since we want to use checked_cast in WavReader.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 22:56:09 +00:00
aluebs@webrtc.org
6f10ae25ea Support block_size greater than chunk_size in Blocker 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 17:28:31 +00:00
andrew@webrtc.org
2510d11c0f Add (safe) uint32_t cast to fix Win64 build. 
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 20:47:42 +00:00
andrew@webrtc.org
048c5029f5 Handle all permissible PCM fields with WavReader. 
I discovered the hard way that Adobe Audition writes an 18 byte format
header with an extra (zero) extension size field. Although:
https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
indicates this field shouldn't exist for PCM, the documentation here:
http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/WAVE.html
doesn't list it as strictly forbidden, only that it _must_ exist for
non-PCM formats.

Audition can write metadata to the file after the audio data, which is
also not forbidden. We now ensure to read only up to the audio payload
length to avoid reading the metadata.

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

Review URL: https://webrtc-codereview.appspot.com/33629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 20:17:21 +00:00
andrew@webrtc.org
40e4767f2b Add NEON intrinsics version for min_max_operations_neon.c 
WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a

Review URL: https://webrtc-codereview.appspot.com/28259004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 06:07:47 +00:00
andrew@webrtc.org
3c31e6e2f9 Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon 
WebRtcSpl_MinValueW16Neon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I33c3853766d7594ed121166288e5325a03b3d6fe

Review URL: https://webrtc-codereview.appspot.com/32639004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 00:24:13 +00:00
henrik.lundin@webrtc.org
ff1a3e36bd Make an AudioEncoder subclass for comfort noise 
BUG=3926
R=bjornv@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 07:29:08 +00:00
andrew@webrtc.org
6fd52f36db Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon. 
WebRtcSpl_DownsampleFastNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ic43f5452eb7e555b998b1d1e79a9e1530be5c948

Review URL: https://webrtc-codereview.appspot.com/24359004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 00:59:48 +00:00
andrew@webrtc.org
ae20d3bbce Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon. 
WebRtcSpl_CrossCorrelationNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I84f9fb953448b62da452ab8dd60e2c0628293587

Review URL: https://webrtc-codereview.appspot.com/30189004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 23:58:39 +00:00
bjornv@webrtc.org
ee43263a50 Cleaned up real_fft APIs due to non-existing NEON code 
There are NEON APIs that are not used. Cleaning that up for better overview.

BUG=3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 16:36:22 +00:00
andrew@webrtc.org
fd4acf6d55 Adding WebRtcSpl_MaxAbsValueW16 intrinsics version 
The modification only uses the unique part of the WebRtcSpl_MaxAbsValue
 function. Pass Spltest.MinMaxOperationTest conformance test on both
 ARMv7 and ARM64. And the single function performance is similar with
 original assembly version on different platforms. If not specified, the
 code is compiled by GCC 4.6. The result is the "X version / C version"
 ratio, and the less is better.

| run 100k times             | cortex-a7 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       32% |        15% |
| Neon intrinsics (GCC 4.6)  |       36% |        37% |
| Neon intrinsics (GCC 4.8)  |       35% |        18% |

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ia2f6822ec58774b401cc440b6751a97e540b5048

Review URL: https://webrtc-codereview.appspot.com/30109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 21:59:02 +00:00
andrew@webrtc.org
1751ee7d32 Remove -flax-vector-conversions flag for ARM NEON building. 
Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.

Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/30239004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 19:36:14 +00:00
aluebs@webrtc.org
8789376cd3 Move ChannelBuffer class to channel_buffer file 
No change in functionallity.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 23:40:25 +00:00
aluebs@webrtc.org
79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands 
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.

A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 20:21:38 +00:00
andrew@webrtc.org
1153322cf8 Build fix for MIPS Android Webview build. 
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-21 16:28:32 +00:00
kjellander@webrtc.org
52bb521b47 Update isolate files for Android APK tests. 
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).

A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.

BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 08:35:05 +00:00
andrew@webrtc.org
5804936052 Add format members to AudioConverter for DCHECKing. 
And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 21:32:14 +00:00
andrew@webrtc.org
f866b2d9f9 Restore the void return type on WriteWavHeader. 
Karl pointed out that the user can check the validity of the input
parameters with CheckWavParameters prior to calling.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/23339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 18:20:06 +00:00
andrew@webrtc.org
a3ed713dad Add a WavReader counterpart to WavWriter. 
Don't bother with a C interface as we currently have no need to call
this from C code. The first use will be in the audioproc tool.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 21:51:03 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
andrew@webrtc.org
508c91683c Build fix for MIPS32R6. 
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 16:26:17 +00:00
andrew@webrtc.org
8328e7c44d Revert "Revert part of r7561, "Refactor audio conversion functions."" 
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/28899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 04:58:14 +00:00
henrik.lundin@webrtc.org
8aa4d2d2cd Creating a C++ wrapper class for VAD 
Also adding a mock. This work is part of an ongoing effort to
encapsulate encoders in AudioEncoder classes. The CNG encoder will also
be implemented as an AudioEncoder class, and will also contain a VAD
C++ wrapper.

BUG=3926
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 13:23:25 +00:00
andrew@webrtc.org
4fc4addc81 Refactor audio conversion functions. 
Use a consistent naming scheme that can be understood at the callsite
without having to refer to documentation.

Remove hacks in AudioBuffer intended to maintain bit-exactness with the
float path. The conversions etc. are now all natural, and instead we
enforce close but not bit-exact output between the two paths.

Output of ApmTest.Process:
https://paste.googleplex.com/5931055831842816

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

Review URL: https://webrtc-codereview.appspot.com/13049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 03:40:10 +00:00
bjornv@webrtc.org
f567095f62 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
 2014-10-29 10:29:16 +00:00
bjornv@webrtc.org
67ca26e087 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
 2014-10-28 13:03:10 +00:00
andrew@webrtc.org
aada86b261 Add a simple AudioConverter class. 
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.

The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.

BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/30779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 18:18:17 +00:00
bjornv@webrtc.org
b69ea9a35a common_audio: Replaced invalid operand in min_max_operations_neon.S" 
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.

BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 14:08:35 +00:00
bjornv@webrtc.org
fab5439112 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
 2014-10-15 04:38:42 +00:00
bjornv@webrtc.org
bc1a4578e0 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
 2014-10-13 14:00:43 +00:00
bjornv@webrtc.org
d4fe824862 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> 
The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 13:01:13 +00:00
bjornv@webrtc.org
3ea35fdb1b 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
 2014-10-09 08:47:02 +00:00
andrew@webrtc.org
4165f7aa22 Add a variable for deciding when to use openmax_dl. 
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 18:01:27 +00:00
andrew@webrtc.org
325cff01b4 Import LappedTransform and friends. 
Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 17:42:18 +00:00