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504 Commits

Author SHA1 Message Date
Laurent Aimar
0888d465c1 Removed down/up mixing support from mpgatofixed32. 
It was incomplete and was creating issues when the audio output provided a
non mono device (close #3272).
 2010-03-02 21:47:38 +01:00
Laurent Aimar
c1e7c9f281 Added support for non native float32, and for float64 (BE/LE) in format.c 2010-02-06 14:40:45 +01:00
Laurent Aimar
420e58dec7 Fixed potential use of uninitialized value (format.c). 2010-02-06 14:40:45 +01:00
Rémi Denis-Courmont
ce25160047 Fix previous commit 2010-01-30 15:59:01 +02:00
Rémi Denis-Courmont
cd3b41614d ugly resampler: use memcpy() 2010-01-30 15:42:52 +02:00
Rémi Denis-Courmont
2e12c3f36d ugly resampler: no need to allocate a buffer when down-sampling 2010-01-30 15:42:51 +02:00
Laurent Aimar
5681ac3636 Removed now useless audio filter float.c 2010-01-30 14:03:14 +01:00
Laurent Aimar
97e5f06070 Implemented fi32 -> fl32/s16 conversion in format.c. 2010-01-30 14:03:14 +01:00
Rémi Denis-Courmont
ff9fe43d8e fixed: add S32N, rewrite FL32 2010-01-30 14:40:32 +02:00
Laurent Aimar
dec529e168 Rewrite of format audio filter. 
It factorizes the common code and implements all conversions between s8, u8,
s16l/b, u16l/b, s24l/b, s32l/b and fl32 (close #3162).
 In cases a direct conversion is not implemented, s16 is used as the middle
format.
 The following direct conversions have been removed: fl32->s24, fl32->u16, u16->fl32.
 2010-01-30 13:12:27 +01:00
Laurent Aimar
899895efc8 Factorized 8->16 bits audio conversions. 2010-01-30 13:12:27 +01:00
Laurent Aimar
f3a2e18cdb Cosmetics (audio format conversions). 2010-01-30 13:12:27 +01:00
Rémi Denis-Courmont
0b764e2a1e Move FL32->FI32 conversion to fixed plugin 2010-01-30 13:58:08 +02:00
Rémi Denis-Courmont
e71c332dc3 fixed: refactor into one module and one submodule 
instead of three submodule and no main module!
 2010-01-30 13:47:26 +02:00
Laurent Aimar
3f7117af00 Fixed heavy memory leaks in bandlimited.c 2010-01-26 21:10:00 +01:00
Laurent Aimar
2cf478feb4 Fixed bandlimited invalid writes when downsampling. 2010-01-24 14:47:51 +01:00
Jean-Baptiste Kempf
88fd7db1ab Update Copyright for simple channel_mixer 2010-01-21 23:52:04 +01:00
Rémi Denis-Courmont
39a6423983 Audio format: requires same sample rate and channels count 
This fixes #3168. The audio_format plugin was ignorantly claiming to
convert the sample rate and channel count, though it obviously did not.
 2010-01-17 20:28:17 +02:00
Rémi Denis-Courmont
cc85d3aef8 bandlimited: avoid large stack allocation (refs #3199) 
In most cases, there is enough space for the 2 old samples in the
input buffer head room. In other cases, we anyway need to memory copy
the whole buffer. So we now use block_Realloc(). This also saves us from
copying every samples when resampling.

Unfortunately, the transcode plugin seems to be feeding crap into the
resampler, thus it still crashes.
 2010-01-17 13:32:57 +02:00
Rémi Denis-Courmont
6fa2a46b78 ...Because we are using pointer arithmetic with float pointers 2010-01-17 13:07:18 +02:00
Rémi Denis-Courmont
430be231b6 Remove all default modules from configure.ac 2010-01-16 15:25:01 +02:00
Rémi Denis-Courmont
22fdf98f4c Remove useless <errno.h> inclusions 2010-01-11 19:08:42 +02:00
Rémi Denis-Courmont
fbd95be806 Move ARM NEON optimizations to arm_neon/ 2010-01-10 18:13:11 +02:00
Jean-Baptiste Kempf
6523ce4ac3 Mono is an audio filter, put it in the SUBCAT_AUDIO_AFILTER cat 2010-01-04 23:38:48 +01:00
Geoffroy Couprie
c766d434b0 Add a bunch of help strings. Feel free to correct them, and add more 2010-01-03 14:19:44 +01:00
Pierre d'Herbemont
b890a56baa Revert "bandlimited: factorize." 
This reverts commit d6b5bc5948.

This wasn't intended from prime time. I guess.
 2009-12-31 17:10:24 +01:00
Rémi Duraffort
d6b5bc5948 bandlimited: factorize. 
Signed-off-by: Pierre d'Herbemont <pdherbemont@free.fr>
 2009-12-31 17:05:15 +01:00
Rémi Denis-Courmont
86cb2cd384 scale tempo: fix typo 2009-12-30 21:50:43 +02:00
Rémi Denis-Courmont
8417358d89 headphone mixer: kill config_Get 2009-12-30 21:50:31 +02:00
Rémi Denis-Courmont
4af1e5a2c4 scale tempo filter: kill config_Get 2009-12-30 20:39:37 +02:00
Rémi Denis-Courmont
d74a0d4f88 band-limited resampler: kill config_Get* 2009-12-30 20:39:37 +02:00
Rémi Denis-Courmont
a607746c84 Parameterized equalizer: kill config_Get 2009-12-30 20:39:37 +02:00
Rémi Denis-Courmont
d2eaf62f53 libdca: kill config_Get 2009-12-30 20:39:36 +02:00
Rémi Denis-Courmont
ff717d8778 liba52: kill config_Get 2009-12-30 20:39:36 +02:00
Rémi Denis-Courmont
b259308b3f mono mixer: kill config_Get 2009-12-30 20:39:36 +02:00
Rémi Denis-Courmont
e9a8f1e165 headphone: kill config_Get 2009-12-30 20:39:35 +02:00
Rémi Denis-Courmont
25232e200b Do not assert memory allocations 2009-12-06 11:58:34 +02:00
JP Dinger
15643af12d Replace argument = realloc( argument, size ); with realloc_or_free() in modules/*, and while at it add assert( argument ) to mark unhandled ENOMEM conditions, also for malloc(). 2009-12-05 22:25:43 +01:00
Rémi Denis-Courmont
d317eb91db bandlimited: check input format more thoroughly (closes #3171) 2009-12-03 23:51:16 +02:00
Rémi Denis-Courmont
20c21aac62 Build fix 2009-11-26 19:33:09 +02:00
Rémi Denis-Courmont
18aed9afae Remove linear resampler 
It is computationally more expensive than the ugly one (and has higher
priority). Yet, in my opinion, it sounds even worse.
 2009-11-26 18:31:56 +02:00
Rémi Denis-Courmont
e83b92d7bb Remove the trivial "resampler" 
The ugly resampler is almost as fast yet way better.
 2009-11-26 18:31:38 +02:00
Rémi Duraffort
4f2f56a58c Source files must not be executable. 2009-11-24 22:17:37 +01:00
Matthias Dahl
dbee243488 fix dts spdif output regression 
dts spdif output was broken by the switch over to the new filter
API.

  * dtstofloat32 should only take control if there is _no_ spdif
    output requested otherwise the decoded stream ends up on the
    spdif device

Signed-off-by: Rémi Denis-Courmont <remi@remlab.net>
 2009-11-24 22:50:15 +02:00
Matthias Dahl
baa1ec0106 fix A/52 spdif output regression 
A/52 spdif output was broken by the switch over to the new filter API.

  * a52tofloat32 should only take control if there is _no_ spdif
    output requested otherwise the decoded stream ends up on the spdif
    device

  * a52tospdif has to set a pts/dts value otherwise the frames get
    (wrongfully) discarded as too late because a wrong pts/dts value
    got evaluated by aout_OutputNextBuffer()

Signed-off-by: Rémi Denis-Courmont <remi@remlab.net>
 2009-11-24 22:50:11 +02:00
Clement Chesnin
8e2f99001d Add audiobargraph plugin (audio part) 
Signed-off-by: Rémi Denis-Courmont <remi@remlab.net>
 2009-11-21 12:45:12 +02:00
Pierre Ynard
772baf9960 Remove useless include 2009-11-10 07:21:32 +01:00
Kaarlo Raiha
66440f9fae bands e.g. should have 10 values, not 9 
Signed-off-by: Ilkka Ollakka <ileoo@iki.fi>
 2009-11-08 16:06:33 +02:00
Rémi Denis-Courmont
2aa61dc55a Rename audio filter2 capability back to audio filter 2009-11-05 23:40:23 +02:00
Rémi Denis-Courmont
694b6814bf Normvol: convert to audio filter2 2009-11-05 23:35:27 +02:00