2005-01-06 01:43:34 +00:00
|
|
|
/*-
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
* Copyright (c) 2005-2009 Ariff Abdullah <ariff@FreeBSD.org>
|
|
|
|
* Portions Copyright (c) Ryan Beasley <ryan.beasley@gmail.com> - GSoC 2006
|
|
|
|
* Copyright (c) 1999 Cameron Grant <cg@FreeBSD.org>
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
* All rights reserved.
|
|
|
|
*
|
|
|
|
* Redistribution and use in source and binary forms, with or without
|
|
|
|
* modification, are permitted provided that the following conditions
|
|
|
|
* are met:
|
|
|
|
* 1. Redistributions of source code must retain the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer.
|
|
|
|
* 2. Redistributions in binary form must reproduce the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer in the
|
|
|
|
* documentation and/or other materials provided with the distribution.
|
|
|
|
*
|
|
|
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
|
|
|
|
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
|
|
|
|
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
|
|
|
|
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
|
|
|
|
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
|
|
|
|
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
|
|
|
|
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
|
|
|
|
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
|
|
|
|
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
|
|
|
|
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
|
|
|
|
* SUCH DAMAGE.
|
|
|
|
*/
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
#ifdef HAVE_KERNEL_OPTION_HEADERS
|
|
|
|
#include "opt_snd.h"
|
|
|
|
#endif
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
#include <dev/sound/pcm/sound.h>
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
#include "feeder_if.h"
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
#define SND_USE_FXDIV
|
|
|
|
#include "snd_fxdiv_gen.h"
|
|
|
|
|
2001-08-23 11:30:52 +00:00
|
|
|
SND_DECLARE_FILE("$FreeBSD$");
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
struct snd_dbuf *
|
2004-01-28 08:02:15 +00:00
|
|
|
sndbuf_create(device_t dev, char *drv, char *desc, struct pcm_channel *channel)
|
2001-03-24 23:10:29 +00:00
|
|
|
{
|
|
|
|
struct snd_dbuf *b;
|
|
|
|
|
2003-02-19 05:47:46 +00:00
|
|
|
b = malloc(sizeof(*b), M_DEVBUF, M_WAITOK | M_ZERO);
|
2001-03-24 23:10:29 +00:00
|
|
|
snprintf(b->name, SNDBUF_NAMELEN, "%s:%s", drv, desc);
|
2002-01-26 22:13:24 +00:00
|
|
|
b->dev = dev;
|
2004-01-28 08:02:15 +00:00
|
|
|
b->channel = channel;
|
2002-01-26 22:13:24 +00:00
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
return b;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_destroy(struct snd_dbuf *b)
|
|
|
|
{
|
2007-03-21 18:17:03 +00:00
|
|
|
sndbuf_free(b);
|
2001-03-24 23:10:29 +00:00
|
|
|
free(b, M_DEVBUF);
|
|
|
|
}
|
|
|
|
|
2003-02-20 17:31:12 +00:00
|
|
|
bus_addr_t
|
|
|
|
sndbuf_getbufaddr(struct snd_dbuf *buf)
|
|
|
|
{
|
|
|
|
return (buf->buf_addr);
|
|
|
|
}
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
static void
|
|
|
|
sndbuf_setmap(void *arg, bus_dma_segment_t *segs, int nseg, int error)
|
|
|
|
{
|
2001-03-24 23:10:29 +00:00
|
|
|
struct snd_dbuf *b = (struct snd_dbuf *)arg;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
|
|
|
|
if (bootverbose) {
|
2004-10-15 03:50:04 +00:00
|
|
|
device_printf(b->dev, "sndbuf_setmap %lx, %lx; ",
|
|
|
|
(u_long)segs[0].ds_addr, (u_long)segs[0].ds_len);
|
|
|
|
printf("%p -> %lx\n", b->buf, (u_long)segs[0].ds_addr);
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
2004-10-15 03:50:04 +00:00
|
|
|
if (error == 0)
|
|
|
|
b->buf_addr = segs[0].ds_addr;
|
|
|
|
else
|
|
|
|
b->buf_addr = 0;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
2001-03-24 23:10:29 +00:00
|
|
|
* Allocate memory for DMA buffer. If the device does not use DMA transfers,
|
|
|
|
* the driver can call malloc(9) and sndbuf_setup() itself.
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
*/
|
2003-02-20 17:31:12 +00:00
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
int
|
2007-04-18 18:26:41 +00:00
|
|
|
sndbuf_alloc(struct snd_dbuf *b, bus_dma_tag_t dmatag, int dmaflags,
|
|
|
|
unsigned int size)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
2004-10-15 03:50:04 +00:00
|
|
|
int ret;
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
b->dmatag = dmatag;
|
2011-03-11 22:42:04 +00:00
|
|
|
b->dmaflags = dmaflags | BUS_DMA_NOWAIT | BUS_DMA_COHERENT;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
b->maxsize = size;
|
|
|
|
b->bufsize = b->maxsize;
|
2004-10-15 03:50:04 +00:00
|
|
|
b->buf_addr = 0;
|
2007-02-01 09:46:03 +00:00
|
|
|
b->flags |= SNDBUF_F_MANAGED;
|
2007-04-18 18:26:41 +00:00
|
|
|
if (bus_dmamem_alloc(b->dmatag, (void **)&b->buf, b->dmaflags,
|
2007-03-21 18:17:03 +00:00
|
|
|
&b->dmamap)) {
|
|
|
|
sndbuf_free(b);
|
2004-10-15 03:50:04 +00:00
|
|
|
return (ENOMEM);
|
2007-03-21 18:17:03 +00:00
|
|
|
}
|
2004-10-15 03:50:04 +00:00
|
|
|
if (bus_dmamap_load(b->dmatag, b->dmamap, b->buf, b->maxsize,
|
|
|
|
sndbuf_setmap, b, 0) != 0 || b->buf_addr == 0) {
|
2007-03-21 18:17:03 +00:00
|
|
|
sndbuf_free(b);
|
2004-10-15 03:50:04 +00:00
|
|
|
return (ENOMEM);
|
|
|
|
}
|
|
|
|
|
|
|
|
ret = sndbuf_resize(b, 2, b->maxsize / 2);
|
|
|
|
if (ret != 0)
|
|
|
|
sndbuf_free(b);
|
2007-03-21 18:17:03 +00:00
|
|
|
|
2004-10-15 03:50:04 +00:00
|
|
|
return (ret);
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
int
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_setup(struct snd_dbuf *b, void *buf, unsigned int size)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
2007-03-21 18:17:03 +00:00
|
|
|
b->flags &= ~SNDBUF_F_MANAGED;
|
2007-02-01 09:46:03 +00:00
|
|
|
if (buf)
|
|
|
|
b->flags |= SNDBUF_F_MANAGED;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
b->buf = buf;
|
|
|
|
b->maxsize = size;
|
|
|
|
b->bufsize = b->maxsize;
|
|
|
|
return sndbuf_resize(b, 2, b->maxsize / 2);
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_free(struct snd_dbuf *b)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
2001-03-24 23:10:29 +00:00
|
|
|
if (b->tmpbuf)
|
|
|
|
free(b->tmpbuf, M_DEVBUF);
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
if (b->shadbuf)
|
|
|
|
free(b->shadbuf, M_DEVBUF);
|
2007-02-01 09:46:03 +00:00
|
|
|
|
2007-03-21 18:17:03 +00:00
|
|
|
if (b->buf) {
|
|
|
|
if (b->flags & SNDBUF_F_MANAGED) {
|
|
|
|
if (b->dmamap)
|
|
|
|
bus_dmamap_unload(b->dmatag, b->dmamap);
|
|
|
|
if (b->dmatag)
|
|
|
|
bus_dmamem_free(b->dmatag, b->buf, b->dmamap);
|
|
|
|
} else
|
|
|
|
free(b->buf, M_DEVBUF);
|
|
|
|
}
|
2001-03-24 23:10:29 +00:00
|
|
|
|
2007-03-21 18:17:03 +00:00
|
|
|
b->tmpbuf = NULL;
|
|
|
|
b->shadbuf = NULL;
|
2001-05-27 14:39:34 +00:00
|
|
|
b->buf = NULL;
|
2007-03-21 18:17:03 +00:00
|
|
|
b->sl = 0;
|
|
|
|
b->dmatag = NULL;
|
|
|
|
b->dmamap = NULL;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
|
|
|
|
2007-06-14 11:15:51 +00:00
|
|
|
#define SNDBUF_CACHE_SHIFT 5
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
int
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_resize(struct snd_dbuf *b, unsigned int blkcnt, unsigned int blksz)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
2007-06-14 11:15:51 +00:00
|
|
|
unsigned int bufsize, allocsize;
|
|
|
|
u_int8_t *tmpbuf;
|
2004-01-28 08:02:15 +00:00
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_LOCK(b->channel);
|
2001-03-24 23:10:29 +00:00
|
|
|
if (b->maxsize == 0)
|
2004-01-28 08:02:15 +00:00
|
|
|
goto out;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
if (blkcnt == 0)
|
|
|
|
blkcnt = b->blkcnt;
|
|
|
|
if (blksz == 0)
|
|
|
|
blksz = b->blksz;
|
2007-06-14 11:15:51 +00:00
|
|
|
if (blkcnt < 2 || blksz < 16 || (blkcnt * blksz) > b->maxsize) {
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_UNLOCK(b->channel);
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
return EINVAL;
|
2004-01-28 08:02:15 +00:00
|
|
|
}
|
2002-01-26 22:13:24 +00:00
|
|
|
if (blkcnt == b->blkcnt && blksz == b->blksz)
|
2004-01-28 08:02:15 +00:00
|
|
|
goto out;
|
2003-04-20 17:08:56 +00:00
|
|
|
|
2007-06-14 11:15:51 +00:00
|
|
|
bufsize = blkcnt * blksz;
|
|
|
|
|
Last (again ?!?) major commit for RELENG_7, featuring total Giant
eradication in/from userland path, countless locking fixes, etc.
- General sleep call through msleep(9) has been converted to condvar(9)
with better consistencies.
- Heavily guard every possible "slow path" entries (open(), close(),
few ioctl()s, sysctls), but once it entering "fast path" (io, interrupt
started), they are free to fly on their own.
- Rearrange locking sequences, resulting better concurrency and
serialization. Large part doesn't even need locking at all, and will be
removed in future. Less clutter, except in few places due to lock
ordering.
- Anonymous mixer object creation/deletion to simplify mixer handling
beyond typical mixer ioctls.
Submitted by: chibis (with modifications)
- Add few mix_[get|set|..] functions to avoid calling mixer_ioctl()
directly using cryptic arguments.
- Locking fixes to avoid possible deadlock with (still under Giant) USB.
- Better simplex/duplex device handling.
- Recover mmap() functionality for recording, which has been lost
since 2.2.x - 3.x (the introduction of newpcm). Full-duplex mmap still
doesn't work (due to VM/page design), but people still can mmap
both by opening each direction separately. mmaped playback is guarantee
to work either way.
- New sysctl: "hw.snd.compat_linux_mmap" to allow PROT_EXEC page
mapping, due to recent changes in linux compatibility layer which
require it. All linux applications that using sound + mmap() (mostly games)
require this to be enabled. Disabled by default.
- Other goodies.. too many, that will increase releng7 shareholder value
and make users of releng6 (and below) cry ;)
* This commit should be atomic. If anything goes wrong (not counting problem
originated from elsewhere), I will not hesitate to revert everything back
within 12 hours. This substantial changes itself not a rocket science
and the process has begun for almost 2 years, and lots of incremental
changes are already in place during that period of time.
* Some issues does occur in snd_emu10kx (note the 'x') due to various
internal locking issues and it is currently being worked on by chibis.
Tested by: chibis (Yuriy Tsibizov), joel, Alexandre Vieira,
many innocent souls...
2007-06-16 03:37:28 +00:00
|
|
|
if (bufsize > b->allocsize ||
|
2007-06-14 11:15:51 +00:00
|
|
|
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
|
|
|
|
allocsize = round_page(bufsize);
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_UNLOCK(b->channel);
|
2007-06-14 11:15:51 +00:00
|
|
|
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_LOCK(b->channel);
|
2007-06-14 11:15:51 +00:00
|
|
|
if (snd_verbose > 3)
|
|
|
|
printf("%s(): b=%p %p -> %p [%d -> %d : %d]\n",
|
|
|
|
__func__, b, b->tmpbuf, tmpbuf,
|
|
|
|
b->allocsize, allocsize, bufsize);
|
|
|
|
if (b->tmpbuf != NULL)
|
|
|
|
free(b->tmpbuf, M_DEVBUF);
|
|
|
|
b->tmpbuf = tmpbuf;
|
|
|
|
b->allocsize = allocsize;
|
|
|
|
} else if (snd_verbose > 3)
|
|
|
|
printf("%s(): b=%p %d [%d] NOCHANGE\n",
|
|
|
|
__func__, b, b->allocsize, b->bufsize);
|
|
|
|
|
2004-01-28 08:02:15 +00:00
|
|
|
b->blkcnt = blkcnt;
|
|
|
|
b->blksz = blksz;
|
2007-06-14 11:15:51 +00:00
|
|
|
b->bufsize = bufsize;
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_reset(b);
|
2004-01-28 08:02:15 +00:00
|
|
|
out:
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_UNLOCK(b->channel);
|
2001-03-24 23:10:29 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
int
|
|
|
|
sndbuf_remalloc(struct snd_dbuf *b, unsigned int blkcnt, unsigned int blksz)
|
|
|
|
{
|
2007-06-14 11:15:51 +00:00
|
|
|
unsigned int bufsize, allocsize;
|
|
|
|
u_int8_t *buf, *tmpbuf, *shadbuf;
|
2002-11-25 17:17:43 +00:00
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
if (blkcnt < 2 || blksz < 16)
|
|
|
|
return EINVAL;
|
|
|
|
|
2002-11-25 17:17:43 +00:00
|
|
|
bufsize = blksz * blkcnt;
|
2001-03-24 23:10:29 +00:00
|
|
|
|
2007-06-14 11:15:51 +00:00
|
|
|
if (bufsize > b->allocsize ||
|
|
|
|
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
|
|
|
|
allocsize = round_page(bufsize);
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_UNLOCK(b->channel);
|
2007-06-14 11:15:51 +00:00
|
|
|
buf = malloc(allocsize, M_DEVBUF, M_WAITOK);
|
|
|
|
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
|
|
|
|
shadbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
CHN_LOCK(b->channel);
|
2007-06-14 11:15:51 +00:00
|
|
|
if (b->buf != NULL)
|
|
|
|
free(b->buf, M_DEVBUF);
|
|
|
|
b->buf = buf;
|
|
|
|
if (b->tmpbuf != NULL)
|
|
|
|
free(b->tmpbuf, M_DEVBUF);
|
|
|
|
b->tmpbuf = tmpbuf;
|
|
|
|
if (b->shadbuf != NULL)
|
|
|
|
free(b->shadbuf, M_DEVBUF);
|
|
|
|
b->shadbuf = shadbuf;
|
|
|
|
if (snd_verbose > 3)
|
|
|
|
printf("%s(): b=%p %d -> %d [%d]\n",
|
|
|
|
__func__, b, b->allocsize, allocsize, bufsize);
|
|
|
|
b->allocsize = allocsize;
|
|
|
|
} else if (snd_verbose > 3)
|
|
|
|
printf("%s(): b=%p %d [%d] NOCHANGE\n",
|
|
|
|
__func__, b, b->allocsize, b->bufsize);
|
2002-11-25 17:17:43 +00:00
|
|
|
|
|
|
|
b->blkcnt = blkcnt;
|
|
|
|
b->blksz = blksz;
|
|
|
|
b->bufsize = bufsize;
|
|
|
|
b->maxsize = bufsize;
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
b->sl = bufsize;
|
2002-11-25 17:17:43 +00:00
|
|
|
|
2004-01-28 08:02:15 +00:00
|
|
|
sndbuf_reset(b);
|
|
|
|
|
2007-06-14 11:15:51 +00:00
|
|
|
return 0;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Zero out space in buffer free area
|
|
|
|
*
|
|
|
|
* This function clears a chunk of @c length bytes in the buffer free area
|
|
|
|
* (i.e., where the next write will be placed).
|
|
|
|
*
|
|
|
|
* @param b buffer context
|
|
|
|
* @param length number of bytes to blank
|
|
|
|
*/
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
void
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_clear(struct snd_dbuf *b, unsigned int length)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
int i;
|
2001-03-05 16:45:38 +00:00
|
|
|
u_char data, *p;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
|
|
|
|
if (length == 0)
|
|
|
|
return;
|
2001-03-05 16:45:38 +00:00
|
|
|
if (length > b->bufsize)
|
|
|
|
length = b->bufsize;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
data = sndbuf_zerodata(b->fmt);
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
i = sndbuf_getfreeptr(b);
|
|
|
|
p = sndbuf_getbuf(b);
|
2001-03-05 16:45:38 +00:00
|
|
|
while (length > 0) {
|
|
|
|
p[i] = data;
|
|
|
|
length--;
|
|
|
|
i++;
|
|
|
|
if (i >= b->bufsize)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
i = 0;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Zap buffer contents, resetting "ready area" fields
|
|
|
|
*
|
|
|
|
* @param b buffer context
|
|
|
|
*/
|
2001-05-27 14:39:34 +00:00
|
|
|
void
|
|
|
|
sndbuf_fillsilence(struct snd_dbuf *b)
|
|
|
|
{
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
if (b->bufsize > 0)
|
|
|
|
memset(sndbuf_getbuf(b), sndbuf_zerodata(b->fmt), b->bufsize);
|
2001-05-27 14:39:34 +00:00
|
|
|
b->rp = 0;
|
|
|
|
b->rl = b->bufsize;
|
|
|
|
}
|
|
|
|
|
2012-01-31 21:46:28 +00:00
|
|
|
void
|
|
|
|
sndbuf_fillsilence_rl(struct snd_dbuf *b, u_int rl)
|
|
|
|
{
|
|
|
|
if (b->bufsize > 0)
|
|
|
|
memset(sndbuf_getbuf(b), sndbuf_zerodata(b->fmt), b->bufsize);
|
|
|
|
b->rp = 0;
|
|
|
|
b->rl = min(b->bufsize, rl);
|
|
|
|
}
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Reset buffer w/o flushing statistics
|
|
|
|
*
|
|
|
|
* This function just zeroes out buffer contents and sets the "ready length"
|
|
|
|
* to zero. This was originally to facilitate minimal playback interruption
|
|
|
|
* (i.e., dropped samples) in SNDCTL_DSP_SILENCE/SKIP ioctls.
|
|
|
|
*
|
|
|
|
* @param b buffer context
|
|
|
|
*/
|
|
|
|
void
|
|
|
|
sndbuf_softreset(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
b->rl = 0;
|
|
|
|
if (b->buf && b->bufsize > 0)
|
|
|
|
sndbuf_clear(b, b->bufsize);
|
|
|
|
}
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
void
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_reset(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
b->hp = 0;
|
|
|
|
b->rp = 0;
|
|
|
|
b->rl = 0;
|
|
|
|
b->dl = 0;
|
|
|
|
b->prev_total = 0;
|
|
|
|
b->total = 0;
|
|
|
|
b->xrun = 0;
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
if (b->buf && b->bufsize > 0)
|
|
|
|
sndbuf_clear(b, b->bufsize);
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
sndbuf_clearshadow(b);
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
u_int32_t
|
|
|
|
sndbuf_getfmt(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->fmt;
|
|
|
|
}
|
|
|
|
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
int
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_setfmt(struct snd_dbuf *b, u_int32_t fmt)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
b->fmt = fmt;
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
b->bps = AFMT_BPS(b->fmt);
|
|
|
|
b->align = AFMT_ALIGN(b->fmt);
|
|
|
|
#if 0
|
|
|
|
b->bps = AFMT_CHANNEL(b->fmt);
|
Whats New:
1. Support wide range sampling rate, as low as 1hz up to int32 max
(which is, insane) through new feeder_rate, multiple precisions
choice (32/64 bit converter). This is indeed, quite insane, but it
does give us more room and flexibility. Plenty sysctl options to
adjust resampling characteristics.
2. Support 24/32 bit pcm format conversion through new, much improved,
simplified and optimized feeder_fmt.
Changes:
1. buffer.c / dsp.c / sound.h
* Support for 24/32 AFMT.
2. feeder_rate.c
* New implementation of sampling rate conversion with 32/64 bit
precision, 1 - int32max hz (which is, ridiculous, yet very
addictive). Much improved / smarter buffer management to not
cause any missing samples at the end of conversion process
* Tunable sysctls for various aspect:
hw.snd.feeder_rate_ratemin - minimum allowable sampling rate
(default to 4000)
hw.snd.feeder_rate_ratemax - maximum allowable sampling rate
(default to 1102500)
hw.snd.feeder_rate_buffersize - conversion buffer size
(default to 8192)
hw.snd.feeder_rate_scaling - scaling / conversion method
(please refer to the source for explaination). Default to
previous implementation type.
3. feeder_fmt.c / sound.h
* New implementation, support for 24/32bit conversion, optimized,
and simplified. Few routines has been removed (8 to xlaw, 16 to
8). It just doesn't make sense.
4. channel.c
* Support for 24/32 AFMT
* Fix wrong xruns increment, causing incorrect underruns statistic
while using vchans.
5. vchan.c
* Support for 24/32 AFMT
* Proper speed / rate detection especially for fixed rate ac97.
User can override it using kernel hint:
hint.pcm.<unit>.vchanrate="xxxx".
Notes / Issues:
* Virtual Channels (vchans)
Enabling vchans can really, really help to solve overrun
issues. This is quite understandable, because it operates
entirely within its own buffering system without relying on
hardware interrupt / state. Even if you don't need vchan,
just enable single channel can help much. Few soundcards
(notably via8233x, sblive, possibly others) have their own
hardware multi channel, and this is unfortunately beyond
vchan reachability.
* The arrival of 24/32 also come with a price. Applications
that can do 24/32bit playback need to be recompiled (notably
mplayer). Use (recompiled) mplayer to experiment / test /
debug this various format using -af format=fmt. Note that
24bit seeking in mplayer is a little bit broken, sometimes
can cause silence or loud static noise. Pausing / seeking
few times can solve this problem.
You don't have to rebuild world entirely for this. Simply
copy /usr/src/sys/sys/soundcard.h to
/usr/include/sys/soundcard.h would suffice. Few drivers also
need recompilation, and this can be done via
/usr/src/sys/modules/sound/.
Support for 24bit hardware playback is beyond the scope of
this changes. That would require spessific hardware driver
changes.
* Don't expect playing 9999999999hz is a wise decision. Be
reasonable. The new feeder_rate implemention provide
flexibility, not insanity. You can easily chew up your CPU
with this kind of mind instability. Please use proper
mosquito repellent device for this obvious cracked brain
attempt. As for testing purposes, you can use (again)
mplayer to generate / play with different sampling rate. Use
something like "mplayer -af resample=192000:0:0 <files>".
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
2005-07-31 16:16:22 +00:00
|
|
|
if (b->fmt & AFMT_16BIT)
|
|
|
|
b->bps <<= 1;
|
|
|
|
else if (b->fmt & AFMT_24BIT)
|
|
|
|
b->bps *= 3;
|
|
|
|
else if (b->fmt & AFMT_32BIT)
|
|
|
|
b->bps <<= 2;
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
#endif
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
unsigned int
|
|
|
|
sndbuf_getspd(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->spd;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_setspd(struct snd_dbuf *b, unsigned int spd)
|
|
|
|
{
|
|
|
|
b->spd = spd;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getalign(struct snd_dbuf *b)
|
|
|
|
{
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
return (b->align);
|
2001-03-24 23:10:29 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getblkcnt(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->blkcnt;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_setblkcnt(struct snd_dbuf *b, unsigned int blkcnt)
|
|
|
|
{
|
|
|
|
b->blkcnt = blkcnt;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getblksz(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->blksz;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_setblksz(struct snd_dbuf *b, unsigned int blksz)
|
|
|
|
{
|
|
|
|
b->blksz = blksz;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getbps(struct snd_dbuf *b)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
return b->bps;
|
|
|
|
}
|
|
|
|
|
|
|
|
void *
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_getbuf(struct snd_dbuf *b)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
return b->buf;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
void *
|
|
|
|
sndbuf_getbufofs(struct snd_dbuf *b, unsigned int ofs)
|
|
|
|
{
|
2002-01-25 02:39:34 +00:00
|
|
|
KASSERT(ofs < b->bufsize, ("%s: ofs invalid %d", __func__, ofs));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return b->buf + ofs;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getsize(struct snd_dbuf *b)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
return b->bufsize;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
unsigned int
|
|
|
|
sndbuf_getmaxsize(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->maxsize;
|
|
|
|
}
|
|
|
|
|
2007-06-14 11:15:51 +00:00
|
|
|
unsigned int
|
|
|
|
sndbuf_getallocsize(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->allocsize;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
unsigned int
|
|
|
|
sndbuf_runsz(struct snd_dbuf *b)
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
{
|
|
|
|
return b->dl;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
void
|
|
|
|
sndbuf_setrun(struct snd_dbuf *b, int go)
|
|
|
|
{
|
|
|
|
b->dl = go? b->blksz : 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
struct selinfo *
|
|
|
|
sndbuf_getsel(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return &b->sel;
|
|
|
|
}
|
|
|
|
|
|
|
|
/************************************************************/
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getxrun(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->xrun;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
2006-07-17 17:43:06 +00:00
|
|
|
sndbuf_setxrun(struct snd_dbuf *b, unsigned int xrun)
|
2001-03-24 23:10:29 +00:00
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
2006-07-17 17:43:06 +00:00
|
|
|
b->xrun = xrun;
|
2001-03-24 23:10:29 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_gethwptr(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->hp;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_sethwptr(struct snd_dbuf *b, unsigned int ptr)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
b->hp = ptr;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getready(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return b->rl;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getreadyptr(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rp >= 0) && (b->rp <= b->bufsize), ("%s: b->rp invalid %d", __func__, b->rp));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return b->rp;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getfree(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return b->bufsize - b->rl;
|
|
|
|
}
|
|
|
|
|
|
|
|
unsigned int
|
|
|
|
sndbuf_getfreeptr(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rp >= 0) && (b->rp <= b->bufsize), ("%s: b->rp invalid %d", __func__, b->rp));
|
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return (b->rp + b->rl) % b->bufsize;
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
u_int64_t
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_getblocks(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->total / b->blksz;
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
u_int64_t
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_getprevblocks(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->prev_total / b->blksz;
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
u_int64_t
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_gettotal(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->total;
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
u_int64_t
|
|
|
|
sndbuf_getprevtotal(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
return b->prev_total;
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
void
|
|
|
|
sndbuf_updateprevtotal(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
SNDBUF_LOCKASSERT(b);
|
|
|
|
|
|
|
|
b->prev_total = b->total;
|
|
|
|
}
|
|
|
|
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
unsigned int
|
|
|
|
sndbuf_xbytes(unsigned int v, struct snd_dbuf *from, struct snd_dbuf *to)
|
|
|
|
{
|
|
|
|
if (from == NULL || to == NULL || v == 0)
|
|
|
|
return 0;
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
return snd_xbytes(v, sndbuf_getalign(from) * sndbuf_getspd(from),
|
|
|
|
sndbuf_getalign(to) * sndbuf_getspd(to));
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
u_int8_t
|
|
|
|
sndbuf_zerodata(u_int32_t fmt)
|
|
|
|
{
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
if (fmt & (AFMT_SIGNED | AFMT_PASSTHROUGH))
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
return (0x00);
|
|
|
|
else if (fmt & AFMT_MU_LAW)
|
|
|
|
return (0x7f);
|
|
|
|
else if (fmt & AFMT_A_LAW)
|
|
|
|
return (0x55);
|
|
|
|
return (0x80);
|
|
|
|
}
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
/************************************************************/
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Acquire buffer space to extend ready area
|
|
|
|
*
|
|
|
|
* This function extends the ready area length by @c count bytes, and may
|
|
|
|
* optionally copy samples from another location stored in @c from. The
|
|
|
|
* counter @c snd_dbuf::total is also incremented by @c count bytes.
|
|
|
|
*
|
|
|
|
* @param b audio buffer
|
|
|
|
* @param from sample source (optional)
|
|
|
|
* @param count number of bytes to acquire
|
|
|
|
*
|
|
|
|
* @retval 0 Unconditional
|
|
|
|
*/
|
update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
|
|
|
int
|
2001-03-24 23:10:29 +00:00
|
|
|
sndbuf_acquire(struct snd_dbuf *b, u_int8_t *from, unsigned int count)
|
|
|
|
{
|
|
|
|
int l;
|
|
|
|
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT(count <= sndbuf_getfree(b), ("%s: count %d > free %d", __func__, count, sndbuf_getfree(b)));
|
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
|
2001-03-24 23:10:29 +00:00
|
|
|
b->total += count;
|
|
|
|
if (from != NULL) {
|
|
|
|
while (count > 0) {
|
2007-03-16 17:13:44 +00:00
|
|
|
l = min(count, sndbuf_getsize(b) - sndbuf_getfreeptr(b));
|
2001-03-24 23:10:29 +00:00
|
|
|
bcopy(from, sndbuf_getbufofs(b, sndbuf_getfreeptr(b)), l);
|
|
|
|
from += l;
|
|
|
|
b->rl += l;
|
|
|
|
count -= l;
|
|
|
|
}
|
|
|
|
} else
|
|
|
|
b->rl += count;
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d, count %d", __func__, b->rl, count));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Dispose samples from channel buffer, increasing size of ready area
|
|
|
|
*
|
|
|
|
* This function discards samples from the supplied buffer by advancing the
|
|
|
|
* ready area start pointer and decrementing the ready area length. If
|
|
|
|
* @c to is not NULL, then the discard samples will be copied to the location
|
|
|
|
* it points to.
|
|
|
|
*
|
|
|
|
* @param b PCM channel sound buffer
|
|
|
|
* @param to destination buffer (optional)
|
|
|
|
* @param count number of bytes to discard
|
|
|
|
*
|
|
|
|
* @returns 0 unconditionally
|
|
|
|
*/
|
2001-03-24 23:10:29 +00:00
|
|
|
int
|
|
|
|
sndbuf_dispose(struct snd_dbuf *b, u_int8_t *to, unsigned int count)
|
|
|
|
{
|
|
|
|
int l;
|
|
|
|
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT(count <= sndbuf_getready(b), ("%s: count %d > ready %d", __func__, count, sndbuf_getready(b)));
|
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
|
2001-03-24 23:10:29 +00:00
|
|
|
if (to != NULL) {
|
|
|
|
while (count > 0) {
|
2007-03-16 17:13:44 +00:00
|
|
|
l = min(count, sndbuf_getsize(b) - sndbuf_getreadyptr(b));
|
2001-03-24 23:10:29 +00:00
|
|
|
bcopy(sndbuf_getbufofs(b, sndbuf_getreadyptr(b)), to, l);
|
|
|
|
to += l;
|
|
|
|
b->rl -= l;
|
|
|
|
b->rp = (b->rp + l) % b->bufsize;
|
|
|
|
count -= l;
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
b->rl -= count;
|
|
|
|
b->rp = (b->rp + count) % b->bufsize;
|
|
|
|
}
|
2001-12-10 08:09:49 +00:00
|
|
|
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d, count %d", __func__, b->rl, count));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
#ifdef SND_DIAGNOSTIC
|
|
|
|
static uint32_t snd_feeder_maxfeed = 0;
|
2011-01-13 18:20:27 +00:00
|
|
|
SYSCTL_UINT(_hw_snd, OID_AUTO, feeder_maxfeed, CTLFLAG_RD,
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
&snd_feeder_maxfeed, 0, "maximum feeder count request");
|
|
|
|
|
|
|
|
static uint32_t snd_feeder_maxcycle = 0;
|
2011-01-13 18:20:27 +00:00
|
|
|
SYSCTL_UINT(_hw_snd, OID_AUTO, feeder_maxcycle, CTLFLAG_RD,
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
&snd_feeder_maxcycle, 0, "maximum feeder cycle");
|
|
|
|
#endif
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
/* count is number of bytes we want added to destination buffer */
|
|
|
|
int
|
|
|
|
sndbuf_feed(struct snd_dbuf *from, struct snd_dbuf *to, struct pcm_channel *channel, struct pcm_feeder *feeder, unsigned int count)
|
|
|
|
{
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
unsigned int cnt, maxfeed;
|
|
|
|
#ifdef SND_DIAGNOSTIC
|
|
|
|
unsigned int cycle;
|
|
|
|
|
|
|
|
if (count > snd_feeder_maxfeed)
|
|
|
|
snd_feeder_maxfeed = count;
|
|
|
|
|
|
|
|
cycle = 0;
|
|
|
|
#endif
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
|
2001-05-27 14:39:34 +00:00
|
|
|
KASSERT(count > 0, ("can't feed 0 bytes"));
|
|
|
|
|
2001-03-24 23:10:29 +00:00
|
|
|
if (sndbuf_getfree(to) < count)
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
return (EINVAL);
|
|
|
|
|
|
|
|
maxfeed = SND_FXROUND(SND_FXDIV_MAX, sndbuf_getalign(to));
|
2001-03-24 23:10:29 +00:00
|
|
|
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
do {
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
cnt = FEEDER_FEED(feeder, channel, to->tmpbuf,
|
|
|
|
min(count, maxfeed), from);
|
|
|
|
if (cnt == 0)
|
|
|
|
break;
|
|
|
|
sndbuf_acquire(to, to->tmpbuf, cnt);
|
|
|
|
count -= cnt;
|
|
|
|
#ifdef SND_DIAGNOSTIC
|
|
|
|
cycle++;
|
|
|
|
#endif
|
|
|
|
} while (count != 0);
|
2001-03-24 23:10:29 +00:00
|
|
|
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
#ifdef SND_DIAGNOSTIC
|
|
|
|
if (cycle > snd_feeder_maxcycle)
|
|
|
|
snd_feeder_maxcycle = cycle;
|
|
|
|
#endif
|
|
|
|
|
|
|
|
return (0);
|
2001-03-24 23:10:29 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
/************************************************************/
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_dump(struct snd_dbuf *b, char *s, u_int32_t what)
|
|
|
|
{
|
|
|
|
printf("%s: [", s);
|
|
|
|
if (what & 0x01)
|
|
|
|
printf(" bufsize: %d, maxsize: %d", b->bufsize, b->maxsize);
|
|
|
|
if (what & 0x02)
|
|
|
|
printf(" dl: %d, rp: %d, rl: %d, hp: %d", b->dl, b->rp, b->rl, b->hp);
|
|
|
|
if (what & 0x04)
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
printf(" total: %ju, prev_total: %ju, xrun: %d", (uintmax_t)b->total, (uintmax_t)b->prev_total, b->xrun);
|
2001-03-24 23:10:29 +00:00
|
|
|
if (what & 0x08)
|
|
|
|
printf(" fmt: 0x%x, spd: %d", b->fmt, b->spd);
|
|
|
|
if (what & 0x10)
|
|
|
|
printf(" blksz: %d, blkcnt: %d, flags: 0x%x", b->blksz, b->blkcnt, b->flags);
|
|
|
|
printf(" ]\n");
|
|
|
|
}
|
|
|
|
|
|
|
|
/************************************************************/
|
|
|
|
u_int32_t
|
|
|
|
sndbuf_getflags(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
return b->flags;
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
sndbuf_setflags(struct snd_dbuf *b, u_int32_t flags, int on)
|
|
|
|
{
|
|
|
|
b->flags &= ~flags;
|
|
|
|
if (on)
|
|
|
|
b->flags |= flags;
|
|
|
|
}
|
|
|
|
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
/**
|
|
|
|
* @brief Clear the shadow buffer by filling with samples equal to zero.
|
|
|
|
*
|
|
|
|
* @param b buffer to clear
|
|
|
|
*/
|
|
|
|
void
|
|
|
|
sndbuf_clearshadow(struct snd_dbuf *b)
|
|
|
|
{
|
|
|
|
KASSERT(b != NULL, ("b is a null pointer"));
|
|
|
|
KASSERT(b->sl >= 0, ("illegal shadow length"));
|
|
|
|
|
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.
General
-------
- Multichannel safe, endian safe, format safe
* Large part of critical pcm filters such as vchan.c, feeder_rate.c,
feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
using them does not cause the pcm data to be converted to 16bit little
endian.
* Macrosses for accessing pcm data safely are defined within sound.h in
the form of PCM_READ_* / PCM_WRITE_*
* Currently, most of them are probably limited for mono/stereo handling,
but the future addition of true multichannel will be much easier.
- Low latency operation
* Well, this require lot more works to do not just within sound driver,
but we're heading towards right direction. Buffer/block sizing within
channel.c is rewritten to calculate precise allocation for various
combination of sample/data/rate size. As a result, applying correct
SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
to what commercial 4front driver do.
* Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
result long delay.
* Eliminate sound truncation if the sound data is too small.
DIY:
1) Download / extract
http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
2) Do a comparison between "cat state*.au > /dev/dsp" and
"for x in state*.au ; do cat $x > /dev/dsp ; done"
- there should be no "perceivable" differences.
Double close for PR kern/31445.
CAVEAT: Low latency come with (unbearable) price especially for poorly
written applications. Applications that trying to act smarter
by requesting (wrong) blocksize/blockcount will suffer the most.
Fixup samples/patches can be found at:
http://people.freebsd.org/~ariff/ports/
- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
due to closer compatibility with 4front driver.
Discussed with: marcus@ (long time ago?)
- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
moved to their own dev sysctl nodes, notably:
hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
Bump __FreeBSD_version.
Driver specific
---------------
- Ditto for sysctls.
- snd_atiixp, snd_es137x, snd_via8233, snd_hda
* Numerous cleanups and fixes.
* _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
This was intended for pure debugging and latency measurement, but proven
good enough in few unexpected and rare cases (such as problematic shared
IRQ with GIANT devices - USB). Polling can be enabled/disabled through
dev.pcm.0.polling. Disabled by default.
- snd_ich
* Fix possible overflow during speed calibration. Delay final
initialization (pcm_setstatus) after calibration finished.
PR: kern/100169
Tested by: Kevin Overman <oberman@es.net>
* Inverted EAPD for few Nec VersaPro.
PR: kern/104715
Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>
Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.
Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
|
|
|
if ((b->shadbuf != NULL) && (b->sl > 0))
|
|
|
|
memset(b->shadbuf, sndbuf_zerodata(b->fmt), b->sl);
|
MFp4 the sound Google Summer of Code project:
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
2006-09-23 20:45:47 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
#ifdef OSSV4_EXPERIMENT
|
|
|
|
/**
|
|
|
|
* @brief Return peak value from samples in buffer ready area.
|
|
|
|
*
|
|
|
|
* Peak ranges from 0-32767. If channel is monaural, most significant 16
|
|
|
|
* bits will be zero. For now, only expects to work with 1-2 channel
|
|
|
|
* buffers.
|
|
|
|
*
|
|
|
|
* @note Currently only operates with linear PCM formats.
|
|
|
|
*
|
|
|
|
* @param b buffer to analyze
|
|
|
|
* @param lpeak pointer to store left peak value
|
|
|
|
* @param rpeak pointer to store right peak value
|
|
|
|
*/
|
|
|
|
void
|
|
|
|
sndbuf_getpeaks(struct snd_dbuf *b, int *lp, int *rp)
|
|
|
|
{
|
|
|
|
u_int32_t lpeak, rpeak;
|
|
|
|
|
|
|
|
lpeak = 0;
|
|
|
|
rpeak = 0;
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @todo fill this in later
|
|
|
|
*/
|
|
|
|
}
|
|
|
|
#endif
|