History log of /frameworks/av/tools/resampler_tools/fir.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
46afbec3743f1d799f185273ff897d1f8e0175dd 04-Nov-2012 Mathias Agopian <mathias@google.com> change how we store the FIR coefficients

The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.

This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).

The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).

Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
/frameworks/av/tools/resampler_tools/fir.cpp
d88a051aff15fdf5c57e1e5a4083bbd9635af3ad 30-Oct-2012 Mathias Agopian <mathias@google.com> fix another issue with generating FIR coefficients

the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.

Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
/frameworks/av/tools/resampler_tools/fir.cpp
b4b75b47c2a4248e60bbc3229d6acc4d5f872431 30-Oct-2012 Mathias Agopian <mathias@google.com> fir a typo that caused up-sampling coefficiens to be wrong

up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.

also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.

Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
/frameworks/av/tools/resampler_tools/fir.cpp
73e90268adf4c9638b8d820a802e5e9a8ebe6597 26-Oct-2012 Pixelflinger <mathias.agopian@gmail.com> improve fir tool: cleanup, better default, bug fixes

- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter

also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
/frameworks/av/tools/resampler_tools/fir.cpp
4cb4f7ce9184de9a221239c28afcf912e7e1ed43 03-Oct-2008 Dan Bornstein <danfuzz@google.com> Manually merge change #111620 from tc3 to mainline, to keep the

automerger from choking on it.

p4 sync
p4 integrate -r -b android_to_tc3 //...@111620,111620
p4 resolve -a
p4 resolve # resolve a couple merge travesties

PRESUBMIT=passed
BUG=1399648
TBR=edheyl
OCL=111902

Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/tools/resampler_tools/fir.cpp
4b61366dedf8536679083004ce0b6ac2b7e52fc2 30-Oct-2007 Mathias Agopian <mathias@google.com> Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now.

Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/tools/resampler_tools/fir.cpp
2a967b3fff07b8711aef41f839ad7521323bb64d 29-Oct-2007 Mathias Agopian <mathias@google.com> A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time.

Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/tools/resampler_tools/fir.cpp
65682fb8e99ab2f1d2ad6a44ed507e78e757ffa9 24-Aug-2007 Mathias Agopian <mathias@google.com> fix a few small typos in the FIR computation

Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/tools/resampler_tools/fir.cpp
4b8a3d8a89814dc3fb365f18d01733e26eb495a1 23-Aug-2007 Mathias Agopian <mathias@google.com> This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.

Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/tools/resampler_tools/fir.cpp