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freebsd-nq/sys/dev/sound/pcm/buffer.c

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Start each of the license/copyright comments with /*-, minor shuffle of lines
2005-01-06 01:43:34 +00:00
/*-
update my email address.
2003-09-07 16:28:03 +00:00
* 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.
*/
#include <dev/sound/pcm/sound.h>
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
#include "feeder_if.h"
many changes: * add new channels to the end of the list so channels used in order of addition * de-globalise definition of struct snddev_info and provide accessor functions where necessary. * move the $FreeBSD$ tag in each .c file into a macro and allow the /dev/sndstat handler to display these when set to maximum verbosity to aid debugging. * allow each device to register its own sndstat handler to reduce the amount of groping sndstat must do in foreign structs.
2001-08-23 11:30:52 +00:00
SND_DECLARE_FILE("$FreeBSD$");
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
struct snd_dbuf *
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
sndbuf_create(device_t dev, char *drv, char *desc, struct pcm_channel *channel)
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
{
struct snd_dbuf *b;
Back out M_* changes, per decision of the TRB. Approved by: trb
2003-02-19 05:47:46 +00:00
b = malloc(sizeof(*b), M_DEVBUF, M_WAITOK | M_ZERO);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
snprintf(b->name, SNDBUF_NAMELEN, "%s:%s", drv, desc);
* improve error handling * be more specific in verbose boot messages * allow the feeder subsystem to veto pcm* attaching if there is an error initialising the root feeder * don't free/malloc a new tmpbuf when resizing a snd_dbuf to the same size as it currently is * store the feeder description in the feeder structure instead of mallocing space for it
2002-01-26 22:13:24 +00:00
b->dev = dev;
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
b->channel = channel;
* improve error handling * be more specific in verbose boot messages * allow the feeder subsystem to veto pcm* attaching if there is an error initialising the root feeder * don't free/malloc a new tmpbuf when resizing a snd_dbuf to the same size as it currently is * store the feeder description in the feeder structure instead of mallocing space for it
2002-01-26 22:13:24 +00:00
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return b;
}
void
sndbuf_destroy(struct snd_dbuf *b)
{
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
sndbuf_free(b);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
free(b, M_DEVBUF);
}
Implement a "sndbuf_getbufaddr" function and use it instead of vtophys(). Reviewed by: orion
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)
{
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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) {
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
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
}
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
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
}
/*
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
*/
Implement a "sndbuf_getbufaddr" function and use it instead of vtophys(). Reviewed by: orion
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
sndbuf_alloc() now accept dmaflags argument which will be forwarded to internal bus_dmammem_alloc() for greater flexibility on setting up DMA / page attributes.
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
{
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
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;
sndbuf_alloc() now accept dmaflags argument which will be forwarded to internal bus_dmammem_alloc() for greater flexibility on setting up DMA / page attributes.
2007-04-18 18:26:41 +00:00
b->dmaflags = dmaflags | BUS_DMA_NOWAIT;
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;
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
2004-10-15 03:50:04 +00:00
b->buf_addr = 0;
Fix huge memory leak within sound buffer (during channel destruction, buffer resizing, etc.) that was here since eon. Free all (unmanaged) allocated buffer through sndbuf_destroy() in case we forgot to call sndbuf_free(). For a managed buffer (mostly hw specific managed buffer), either provide CHANNEL_FREE() method with appropriate return value to invoke semi-automatic sndbuf_free() or simply do it on their own. If everything is failed, sndbuf_destroy() will come to the rescue as a final measure. MFC after: 3 days
2007-02-01 09:46:03 +00:00
b->flags |= SNDBUF_F_MANAGED;
sndbuf_alloc() now accept dmaflags argument which will be forwarded to internal bus_dmammem_alloc() for greater flexibility on setting up DMA / page attributes.
2007-04-18 18:26:41 +00:00
if (bus_dmamem_alloc(b->dmatag, (void **)&b->buf, b->dmaflags,
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
&b->dmamap)) {
sndbuf_free(b);
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
2004-10-15 03:50:04 +00:00
return (ENOMEM);
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
}
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
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) {
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
sndbuf_free(b);
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
2004-10-15 03:50:04 +00:00
return (ENOMEM);
}
ret = sndbuf_resize(b, 2, b->maxsize / 2);
if (ret != 0)
sndbuf_free(b);
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
Plug possible memory leak in sound DMA buffer handling. It also changes return code to ENOMEM in case of allocation failure. Approved by: jake (mentor), scottl (co-mentor) Reviewed by: truckman, matk
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
{
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
b->flags &= ~SNDBUF_F_MANAGED;
Fix huge memory leak within sound buffer (during channel destruction, buffer resizing, etc.) that was here since eon. Free all (unmanaged) allocated buffer through sndbuf_destroy() in case we forgot to call sndbuf_free(). For a managed buffer (mostly hw specific managed buffer), either provide CHANNEL_FREE() method with appropriate return value to invoke semi-automatic sndbuf_free() or simply do it on their own. If everything is failed, sndbuf_destroy() will come to the rescue as a final measure. MFC after: 3 days
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
{
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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);
Fix huge memory leak within sound buffer (during channel destruction, buffer resizing, etc.) that was here since eon. Free all (unmanaged) allocated buffer through sndbuf_destroy() in case we forgot to call sndbuf_free(). For a managed buffer (mostly hw specific managed buffer), either provide CHANNEL_FREE() method with appropriate return value to invoke semi-automatic sndbuf_free() or simply do it on their own. If everything is failed, sndbuf_destroy() will come to the rescue as a final measure. MFC after: 3 days
2007-02-01 09:46:03 +00:00
Memory leak killing spree, mostly bus_dma(9) related.
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);
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
Memory leak killing spree, mostly bus_dma(9) related.
2007-03-21 18:17:03 +00:00
b->tmpbuf = NULL;
b->shadbuf = NULL;
don't erase info in sndbuf_setup() set free'd pointers to NULL in sndbuf_free() add a new function
2001-05-27 14:39:34 +00:00
b->buf = NULL;
Memory leak killing spree, mostly bus_dma(9) related.
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
}
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
{
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
unsigned int bufsize, allocsize;
u_int8_t *tmpbuf;
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
chn_lock(b->channel);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
if (b->maxsize == 0)
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
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;
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
if (blkcnt < 2 || blksz < 16 || (blkcnt * blksz) > b->maxsize) {
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +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;
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
}
* improve error handling * be more specific in verbose boot messages * allow the feeder subsystem to veto pcm* attaching if there is an error initialising the root feeder * don't free/malloc a new tmpbuf when resizing a snd_dbuf to the same size as it currently is * store the feeder description in the feeder structure instead of mallocing space for it
2002-01-26 22:13:24 +00:00
if (blkcnt == b->blkcnt && blksz == b->blksz)
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
goto out;
Additional malloc failure checks.
2003-04-20 17:08:56 +00:00
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
bufsize = blkcnt * blksz;
if (b->tmpbuf == NULL || bufsize > b->allocsize ||
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
allocsize = round_page(bufsize);
chn_unlock(b->channel);
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
chn_lock(b->channel);
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);
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
b->blkcnt = blkcnt;
b->blksz = blksz;
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
b->bufsize = bufsize;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
sndbuf_reset(b);
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
out:
chn_unlock(b->channel);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return 0;
}
int
sndbuf_remalloc(struct snd_dbuf *b, unsigned int blkcnt, unsigned int blksz)
{
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
unsigned int bufsize, allocsize;
u_int8_t *buf, *tmpbuf, *shadbuf;
various fixes to eliminate locking warnings Approved by: re Reviewed by: orion
2002-11-25 17:17:43 +00:00
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
if (blkcnt < 2 || blksz < 16)
return EINVAL;
various fixes to eliminate locking warnings Approved by: re Reviewed by: orion
2002-11-25 17:17:43 +00:00
bufsize = blksz * blkcnt;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
if (bufsize > b->allocsize ||
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
allocsize = round_page(bufsize);
chn_unlock(b->channel);
buf = malloc(allocsize, M_DEVBUF, M_WAITOK);
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
shadbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
chn_lock(b->channel);
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);
various fixes to eliminate locking warnings Approved by: re Reviewed by: orion
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;
various fixes to eliminate locking warnings Approved by: re Reviewed by: orion
2002-11-25 17:17:43 +00:00
Change KASSERT() in feed_vchan16() into an explicit test and call to panic() so that the buffer overflow just beyond this point is always caught, even when the code is not compiled with INVARIANTS. Change chn_setblocksize() buffer reallocation code to attempt to avoid the feed_vchan16() buffer overflow by attempting to always keep the bufsoft buffer at least as large as the bufhard buffer. Print a diagnositic message Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE() if our best attempts fail. If feed_vchan16() were to be called by the interrupt handler while locks are dropped in chn_setblocksize() to increase the size bufsoft to match the size of bufhard, the panic() code in feed_vchan16() will be triggered. If the diagnostic message is printed, it is a warning that a panic is possible if the system were to see events in an "unlucky" order. Change the locking code to avoid the need for MTX_RECURSIVE mutexes. Add the MTX_DUPOK option to the channel mutexes and change the locking sequence to always lock the parent channel before its children to avoid the possibility of deadlock. Actually implement locking assertions for the channel mutexes and fix the problems found by the resulting assertion violations. Clean up the locking code in dsp_ioctl(). Allocate the channel buffers using the malloc() M_WAITOK option instead of M_NOWAIT so that buffer allocation won't fail. Drop locks across the malloc() calls. Add/modify KASSERTS() in attempt to detect problems early. Abuse layering by adding a pointer to the snd_dbuf structure that points back to the pcm_channel that owns it. This allows sndbuf_resize() to do proper locking without having to change the its API, which is used by the hardware drivers. Don't dereference a NULL pointer when setting hw.snd.maxautovchans if a hardware driver is not loaded. Noticed by Ryan Sommers <ryans at gamersimpact.com>. Tested by: Stefan Ehmann <shoesoft AT gmx.net> Tested by: matk (Mathew Kanner) Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
2004-01-28 08:02:15 +00:00
sndbuf_reset(b);
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
MFS: don't pagefault in sndbuf_clear
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;
MFS: don't pagefault in sndbuf_clear
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
i = sndbuf_getfreeptr(b);
p = sndbuf_getbuf(b);
MFS: don't pagefault in sndbuf_clear
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
*/
don't erase info in sndbuf_setup() set free'd pointers to NULL in sndbuf_free() add a new function
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);
don't erase info in sndbuf_setup() set free'd pointers to NULL in sndbuf_free() add a new function
2001-05-27 14:39:34 +00:00
b->rp = 0;
b->rl = 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
/**
* @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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
b->bps = 1;
b->bps <<= (b->fmt & AFMT_STEREO)? 1 : 0;
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;
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;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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)
{
static int align[] = {0, 1, 1, 2, 2, 2, 2, 3};
return align[b->bps - 1];
}
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 *
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
void *
sndbuf_getbufofs(struct snd_dbuf *b, unsigned int ofs)
{
a buffer offset equal to the buffer size is illegal too, fix assertion in sndbuf_getbufofs()
2002-01-25 02:39:34 +00:00
KASSERT(ofs < b->bufsize, ("%s: ofs invalid %d", __func__, ofs));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
unsigned int
sndbuf_getmaxsize(struct snd_dbuf *b)
{
return b->maxsize;
}
Buffer optimization and locking cleanup. Don't resize/malloc unless it is really necessary to ease down unlock/lock sequence.
2007-06-14 11:15:51 +00:00
unsigned int
sndbuf_getallocsize(struct snd_dbuf *b)
{
return b->allocsize;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
Rename some variables. This fixes some (but not all) problems on the way for WARNS > 2 cleanlyness. Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2006-07-17 17:43:06 +00:00
sndbuf_setxrun(struct snd_dbuf *b, unsigned int xrun)
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
{
SNDBUF_LOCKASSERT(b);
Rename some variables. This fixes some (but not all) problems on the way for WARNS > 2 cleanlyness. Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2006-07-17 17:43:06 +00:00
b->xrun = xrun;
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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);
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
2001-12-10 08:09:49 +00:00
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return b->rl;
}
unsigned int
sndbuf_getreadyptr(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
2001-12-10 08:09:49 +00:00
KASSERT((b->rp >= 0) && (b->rp <= b->bufsize), ("%s: b->rp invalid %d", __func__, b->rp));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return b->rp;
}
unsigned int
sndbuf_getfree(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
2001-12-10 08:09:49 +00:00
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return b->bufsize - b->rl;
}
unsigned int
sndbuf_getfreeptr(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
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));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return (b->rp + b->rl) % b->bufsize;
}
unsigned int
sndbuf_getblocks(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->total / b->blksz;
}
unsigned int
sndbuf_getprevblocks(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->prev_total / b->blksz;
}
unsigned int
sndbuf_gettotal(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->total;
}
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
snd_xbytes(unsigned int v, unsigned int from, unsigned int to)
{
unsigned int w, x, y;
if (from == to)
return v;
if (from == 0 || to == 0 || v == 0)
return 0;
x = from;
y = to;
while (y != 0) {
w = x % y;
x = y;
y = w;
}
from /= x;
to /= x;
return (unsigned int)(((u_int64_t)v * to) / from);
}
unsigned int
sndbuf_xbytes(unsigned int v, struct snd_dbuf *from, struct snd_dbuf *to)
{
if (from == NULL || to == NULL || v == 0)
return 0;
return snd_xbytes(v, sndbuf_getbps(from) * sndbuf_getspd(from),
sndbuf_getbps(to) * sndbuf_getspd(to));
}
u_int8_t
sndbuf_zerodata(u_int32_t fmt)
{
if (fmt & AFMT_SIGNED)
return (0x00);
else if (fmt & AFMT_MU_LAW)
return (0x7f);
else if (fmt & AFMT_A_LAW)
return (0x55);
return (0x80);
}
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
sndbuf_acquire(struct snd_dbuf *b, u_int8_t *from, unsigned int count)
{
int l;
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
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));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
b->total += count;
if (from != NULL) {
while (count > 0) {
[stage: 2/9] Use inlined min() rather than MIN() macross.
2007-03-16 17:13:44 +00:00
l = min(count, sndbuf_getsize(b) - sndbuf_getfreeptr(b));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
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));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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
*/
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
int
sndbuf_dispose(struct snd_dbuf *b, u_int8_t *to, unsigned int count)
{
int l;
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
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));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
if (to != NULL) {
while (count > 0) {
[stage: 2/9] Use inlined min() rather than MIN() macross.
2007-03-16 17:13:44 +00:00
l = min(count, sndbuf_getsize(b) - sndbuf_getreadyptr(b));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
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;
}
Update to C99, s/__FUNCTION__/__func__/, also don't use ANSI string concatenation.
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));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return 0;
}
/* 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)
{
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 cnt;
don't erase info in sndbuf_setup() set free'd pointers to NULL in sndbuf_free() add a new function
2001-05-27 14:39:34 +00:00
KASSERT(count > 0, ("can't feed 0 bytes"));
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
if (sndbuf_getfree(to) < count)
return EINVAL;
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 {
cnt = FEEDER_FEED(feeder, channel, to->tmpbuf, count, from);
Last major commit and updates for RELENG_7: - Rework the entire pcm_channel structure: * Remove rarely used link placeholder, instead, make each pcm_channel as head/link of each own/each other. Unlock - Lock sequence due to sleep malloc has been reduced. * Implement "busy" queue which will contain list of busy/active channels. This greatly reduce locking contention for example while servicing interrupt for hardware with many channels or when virtual channels reach its 256 peak channels. - So I heard you like v chan ... O RLY? Welcome to Virtual **Record** Channels (vrec, rec vchans, vchans for recording, Rec-Chan, you decide), the ultimate solutions for your nagging O_RDWR full-duplex wannabe (note: flash plugins) monopolizing single record channel causing EBUSY. Vrec works exactly like Vchans (or, should I rename it to "Vplay" :) , except that it operates on the opposite direction (recording). Up to 256 vrecs (like vchans) are possible. Notes: * Relocate dev.pcm.%d.{vchans,vchanformat,vchanrate} to each of its respective node/direction: dev.pcm.%d.play.* for "play" (cdev = dsp%d.vp%d) dev.pcm.%d.rec.* for "record" (cdev = dsp%d.vr%d) * Don't expect that it will magically give you ability to split "recording source" (eg: 1 channel for cdrom, 1 channel for mic, etc). Just admit that you only have a *single* recording source / channel. Please bug your hardware vendor instead :) - Bump maxautovchans from 4 to 16. For a full-fledged multimedia desktop/workstation with too many soundservers installed (esound, artsd, jackd, pulse/polypaudio, ding-dong pling plong mudkip fuh fuh, etc), 4 seems inadequate. There will be no memory penalty here, since virtual channels are allocate only by demand. - Nuke/Rework the entire statically created cdev entries. Everything is clonable through snd own clone manager which designed to withstand many kind of abusive devfs droids such as: * while : ; do /bin/test -e /dev/dsp ; done * jot 16777216 0 | while read x ; do ls /dev/dsp0.$x ; done * hundreds (could be thousands) concurrent threads/process opening "/dev/dsp" (previously, this might result EBUSY even with just 3 contesting threads/procs). o Reusable clone objects (instead of creating new one like there's no tomorrow) after certain expiration deadline. The clone allocator will decide whether to reuse, share, or creating new clone. o Automatic garbage collector. - Dynamic unit magic allocator. Maximum attached soundcards can be tuned using tunable "hw.snd.maxunit" (Default to 512). Minimum is 16, and maximum is 2048. - ..other fixes, mostly related to concurrency issues. joel@ will do the manpage updates on sound(4). Have fun.
2007-05-31 18:43:33 +00:00
if (cnt) {
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
sndbuf_acquire(to, to->tmpbuf, cnt);
Last major commit and updates for RELENG_7: - Rework the entire pcm_channel structure: * Remove rarely used link placeholder, instead, make each pcm_channel as head/link of each own/each other. Unlock - Lock sequence due to sleep malloc has been reduced. * Implement "busy" queue which will contain list of busy/active channels. This greatly reduce locking contention for example while servicing interrupt for hardware with many channels or when virtual channels reach its 256 peak channels. - So I heard you like v chan ... O RLY? Welcome to Virtual **Record** Channels (vrec, rec vchans, vchans for recording, Rec-Chan, you decide), the ultimate solutions for your nagging O_RDWR full-duplex wannabe (note: flash plugins) monopolizing single record channel causing EBUSY. Vrec works exactly like Vchans (or, should I rename it to "Vplay" :) , except that it operates on the opposite direction (recording). Up to 256 vrecs (like vchans) are possible. Notes: * Relocate dev.pcm.%d.{vchans,vchanformat,vchanrate} to each of its respective node/direction: dev.pcm.%d.play.* for "play" (cdev = dsp%d.vp%d) dev.pcm.%d.rec.* for "record" (cdev = dsp%d.vr%d) * Don't expect that it will magically give you ability to split "recording source" (eg: 1 channel for cdrom, 1 channel for mic, etc). Just admit that you only have a *single* recording source / channel. Please bug your hardware vendor instead :) - Bump maxautovchans from 4 to 16. For a full-fledged multimedia desktop/workstation with too many soundservers installed (esound, artsd, jackd, pulse/polypaudio, ding-dong pling plong mudkip fuh fuh, etc), 4 seems inadequate. There will be no memory penalty here, since virtual channels are allocate only by demand. - Nuke/Rework the entire statically created cdev entries. Everything is clonable through snd own clone manager which designed to withstand many kind of abusive devfs droids such as: * while : ; do /bin/test -e /dev/dsp ; done * jot 16777216 0 | while read x ; do ls /dev/dsp0.$x ; done * hundreds (could be thousands) concurrent threads/process opening "/dev/dsp" (previously, this might result EBUSY even with just 3 contesting threads/procs). o Reusable clone objects (instead of creating new one like there's no tomorrow) after certain expiration deadline. The clone allocator will decide whether to reuse, share, or creating new clone. o Automatic garbage collector. - Dynamic unit magic allocator. Maximum attached soundcards can be tuned using tunable "hw.snd.maxunit" (Default to 512). Minimum is 16, and maximum is 2048. - ..other fixes, mostly related to concurrency issues. joel@ will do the manpage updates on sound(4). Have fun.
2007-05-31 18:43:33 +00:00
count -= cnt;
}
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
} while (count && cnt);
mega-commit. this introduces a new buffering mechanism which results in dramatic simplification of the channel manager. as several structures have changed, we take the opportunity to move their definitions into the source files where they are used, make them private and de-typedef them. the sound drivers are updated to use snd_setup_intr instead of bus_setup_intr, and to comply with the de-typedefed structures. the ac97, mixer and channel layers have been updated with finegrained locking, as have some drivers- not all though. the rest will follow soon.
2001-03-24 23:10:29 +00:00
return 0;
}
/************************************************************/
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)
printf(" total: %d, prev_total: %d, xrun: %d", b->total, b->prev_total, b->xrun);
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
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