| 505fd3025855f424353ae084495e1855522cf65b |
|
17-Feb-2016 |
Glenn Kasten <gkasten@google.com> |
Disable ARM assembly optimization of legacy linear resampler
It has magic numbers in the asm() that no longer work,
isn't used any more and not really much faster than the C version.
Bug: 13102576
Change-Id: I56398a749ee0a45f31de9317e0a57e38d7b68a69
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| d79072e9dff59f767cce2cda1caab80ce5a0815b |
|
06-Jan-2016 |
Glenn Kasten <gkasten@google.com> |
Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 6b3b7e304e0f8f167241b2c75f1eb04a9ef192ec |
|
29-Mar-2015 |
Andy Hung <hunga@google.com> |
Return number of frames output from resample method
Change-Id: Ic297e2ed59839f1788c83e099ef1a9e4af29591f
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 8a9737ce64a53f06a2c1774b4ea025897c63bff1 |
|
03-Dec-2014 |
Elliott Hughes <enh@google.com> |
__ARM_HAVE_HALFWORD_MULTIPLY is always true for __arm__.
And this is the only user of that macro...
Bug: 18556103
Change-Id: I1b515a54ac8530b150f4dd93070db60c701a5603
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 5e58b0abe5b6c8f5bd96a8f78bbeeeb4d3892020 |
|
24-Jun-2014 |
Andy Hung <hunga@google.com> |
Add floating point volume handling to AudioMixer
Use floating point volume in AudioMixer mixing when floating
point input is used with the new mixer engine.
AudioResampler is updated to take floating point volume to match.
Both legacy integer and floating point mixer engines work.
For now, integer volume is used when the new mixer engine
runs in integer input mode, for backward compatibility with
the legacy mixer. The new mixer engine will generally run in
floating point input mode. When the legacy path is removed,
the integer volumes will be removed.
Change-Id: I79e80c292ae7c8b8bdd0aa371a1b2c3a1b618290
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 3348e36c51e91e78020bcc6578eda83d97c31bec |
|
07-Jul-2014 |
Andy Hung <hunga@google.com> |
Replace int bitDepth with audio_format_t in Resampler
Remove mBitDepth from class (not used).
Replace with audio_format_t in factory method to distinguish
between float and pcm 16-bit.
Change-Id: I166860796c68285077ef4458d8758d19b82523f9
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 075abae2a954bf3edf18ad1705c2c0f188454ae0 |
|
10-Apr-2014 |
Andy Hung <hunga@google.com> |
Add and enable multichannel for audio resampler
Change-Id: I2b86fb73d70abc4c456f7567270a888086b301d4
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 771386e6e6e79697e2d839ef0f25a242946ba1e5 |
|
09-Apr-2014 |
Andy Hung <hunga@google.com> |
Add and enable floating point option for audio resampler
Can be tested with test-resample.
Change-Id: I8339846d7c647444b6025d33cfa145d5d3658121
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 24781fff62a4cf7279d3dac83c33e2ac612712ba |
|
19-Feb-2014 |
Andy Hung <hunga@google.com> |
Fix resampler to allow output of single frame
Bug: 13073201
Change-Id: If7818a3389a191a37277bbd8e96a59ef8ce68509
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 01d3acba9de861cb2b718338e787cff3566fc5ec |
|
06-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix clang warnings in AudioFlinger
Change-Id: I0fa61025c979709ad7d655bc717df5f194b6089e
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 4a25471795b05ffcfcc079e9f9cb5ed549023d14 |
|
05-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 5ef74dce: am d66cd743: Merge "audioflinger: conform inline ASM to UAL"
* commit '5ef74dce64a1b70534547c044a6d15533f4a9dca':
audioflinger: conform inline ASM to UAL
|
| 5f51ade2c290e239a125dc88943b240e1105fd97 |
|
04-Feb-2014 |
synergy dev <synergye@codefi.re> |
audioflinger: conform inline ASM to UAL
Clang requires some inline ASM to conform to the UAL standards (Unified Assembler Language).
This fixes a small issue in this inline asm to allow building.
Change-Id: Ifd9b1814343ab5ade636b9401a21d575559dac16
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 9e0308c03d4e76d3146cbb6e30aeb3ac03f05cf5 |
|
30-Jan-2014 |
Andy Hung <hunga@google.com> |
Enable new dynamic audio resampler as default
The default may still be overridden by af.resampler.quality
in the local or build properties file. For the build properties
check the relevant device.mk files.
In the absence of being overridden, the 44.1 to 48kHz and
48 to 44.1kHz conversion will use the DYN_MED_QUALITY resamplers
and other conversion will use the DYN_LOW_QUALITY resamplers.
Change-Id: Icf9a4af5422cddc445e1904eef93984b0abe6a44
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| ee931ff7d6620e5705f4dfba901fdb03fa4a35fd |
|
28-Jan-2014 |
Andy Hung <hunga@google.com> |
Remove warnings in Audio mixer resampler compilation.
Change-Id: Ie68766142796772a54a90afdea188d0a5a78d4ac
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 86eae0e5931103e040ac2cdd023ef5db252e09f6 |
|
09-Dec-2013 |
Andy Hung <hunga@google.com> |
Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 6e2ebe97f2ad0a21907f20f9ee644c4eacbb7a40 |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use curly braces in 'if' to make it easier to add logs
Change-Id: I58b33fefdd8bf703647414157a99a3223be3531c
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 51ba3c7c9a4a238cdef57ad75a2d8f406ffd4515 |
|
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 93d0767a8a9ee9d649eea9afac59f778e29a6a54 |
|
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| f1b2a9b65a904b10f91482cc0389bea1f66a891f |
|
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| c4640c9eef850bb1c754bd6b477f1cc8350c6081 |
|
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| a6d41334d25ffde12484eb28301352560a063ef6 |
|
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Integrate improved coefficient sinc resampler: VHQ
Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| ac6020508acedd316391dee42329040bf45f8d90 |
|
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Integrate improved coefficient sinc resampler: VHQ
Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 41dfd129a181a5b0b62628733cf96e0ce11d64d7 |
|
17-Jan-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger: use resample coefficients from audio-resampler library.
-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 76b111685010e1fea7c0a865c038aee35507fde4 |
|
17-Jan-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger: use resample coefficients from audio-resampler library.
-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| e53b9ead781c36e96d6b6f012ddffc93a3d80f0d |
|
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 4ff14bae91075eb274eb1c2975982358946e7e63 |
|
09-Feb-2012 |
John Grossman <johngro@google.com> |
Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 3a144d08b713e3c0f6b7b8e95bc42cef5886f4fa |
|
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Disable HQ resamplers for now until qualified"
|
| cdf2158f3b9498d6cd0eb228d8bee16e32399e16 |
|
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Disable HQ resamplers for now until qualified
This saves about 6500 bytes.
Change-Id: I87102fe561c95c19c9e615dea3de914f96639257
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 90bebef5669a9385c706b042d146a31dca2e5d9b |
|
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
No newline or space at end of ALOG format string
Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| d5e54f7a36daedc3b2a642d1499c262da04e6280 |
|
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
mFormat is unused in resampler
mClientTid is unused
local variable pid is unused in dump
Change-Id: Ib156e38029366620bfeff2a13e73471867155a5b
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2 |
|
09-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE
See https://android-git.corp.google.com/g/157519
Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| b8a805261bf0282e992d3608035e47d05a898710 |
|
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| c23e2f2464eb3748599d47af7d8986b856f3c179 |
|
17-Nov-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 re-enable ARM opts but disable inline
PIE did not work together with inlining. This change combines (almost
all of) the performance benefits of ARM optimizations together with PIE.
Change-Id: I4594d33ae5a0a7bac327ae08e30fb35343a06256
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 305f4b6dabe500b97062e72049a296d0699243d7 |
|
28-Oct-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 temporary workaround
Change-Id: I21d91d7a24df7bb6e7fc3d0fbc4786d55391fc0e
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| eb8b914ad9d4331e1cdf4346731770ce69fd0e77 |
|
16-Sep-2011 |
Nick Kralevich <nnk@google.com> |
AudioResampler: use relative instead of absolute labels
This change is needed to allow Android to compile with -fPIE
Bug: 5328392
Change-Id: I84d947975776800a7b79c6ac75a881af461a631c
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 4d0243980123378e9dd69022bf4ddca729d84126 |
|
12-Apr-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 4430670f: am ca7ad44a: Merge "audioflinger: Enable ARMv5TE optimized resampler"
* commit '4430670f48f75661293371dab8db55865f5f56e0':
audioflinger: Enable ARMv5TE optimized resampler
|
| 0c0a1c0c37dbd2646a732da706d6777283c83e44 |
|
06-Apr-2011 |
Jim Huang <jim.huang@linaro.org> |
audioflinger: Enable ARMv5TE optimized resampler
Previously, the optimized asm option is only enabled when
__ARM_ARCH_5E__ is defined, which is assigned in armv5te.mk
rather than armv7-a series targets. This patch checks the ARM CPU
feature about half-word multiply instructions to enable ARMv5TE
resampler optimization routines properly.
Change-Id: I4c5a5d8c932416f23bedb0b389db958349f21ea4
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 243f5f91755c01614a8cafe90b0806396e22d553 |
|
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3479042.
The problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.
The fix consists in implementing a reset function in the resampler.
A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).
Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
/frameworks/av/services/audioflinger/AudioResampler.cpp
|
| 65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 |
|
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
/frameworks/av/services/audioflinger/AudioResampler.cpp
|