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

794 lines
17 KiB
C
Raw Normal View History

/*-
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
* Copyright (c) 2005-2009 Ariff Abdullah <ariff@FreeBSD.org>
* Portions Copyright (c) Ryan Beasley <ryan.beasley@gmail.com> - GSoC 2006
* Copyright (c) 1999 Cameron Grant <cg@FreeBSD.org>
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
#include "feeder_if.h"
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#define SND_USE_FXDIV
#include "snd_fxdiv_gen.h"
SND_DECLARE_FILE("$FreeBSD$");
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)
{
struct snd_dbuf *b;
b = malloc(sizeof(*b), M_DEVBUF, M_WAITOK | M_ZERO);
snprintf(b->name, SNDBUF_NAMELEN, "%s:%s", drv, desc);
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;
return b;
}
void
sndbuf_destroy(struct snd_dbuf *b)
{
sndbuf_free(b);
free(b, M_DEVBUF);
}
bus_addr_t
sndbuf_getbufaddr(struct snd_dbuf *buf)
{
return (buf->buf_addr);
}
static void
sndbuf_setmap(void *arg, bus_dma_segment_t *segs, int nseg, int error)
{
struct snd_dbuf *b = (struct snd_dbuf *)arg;
if (bootverbose) {
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);
}
if (error == 0)
b->buf_addr = segs[0].ds_addr;
else
b->buf_addr = 0;
}
/*
* Allocate memory for DMA buffer. If the device does not use DMA transfers,
* the driver can call malloc(9) and sndbuf_setup() itself.
*/
int
sndbuf_alloc(struct snd_dbuf *b, bus_dma_tag_t dmatag, int dmaflags,
unsigned int size)
{
int ret;
b->dmatag = dmatag;
b->dmaflags = dmaflags | BUS_DMA_NOWAIT | BUS_DMA_COHERENT;
b->maxsize = size;
b->bufsize = b->maxsize;
b->buf_addr = 0;
b->flags |= SNDBUF_F_MANAGED;
if (bus_dmamem_alloc(b->dmatag, (void **)&b->buf, b->dmaflags,
&b->dmamap)) {
sndbuf_free(b);
return (ENOMEM);
}
if (bus_dmamap_load(b->dmatag, b->dmamap, b->buf, b->maxsize,
sndbuf_setmap, b, 0) != 0 || b->buf_addr == 0) {
sndbuf_free(b);
return (ENOMEM);
}
ret = sndbuf_resize(b, 2, b->maxsize / 2);
if (ret != 0)
sndbuf_free(b);
return (ret);
}
int
sndbuf_setup(struct snd_dbuf *b, void *buf, unsigned int size)
{
b->flags &= ~SNDBUF_F_MANAGED;
if (buf)
b->flags |= SNDBUF_F_MANAGED;
b->buf = buf;
b->maxsize = size;
b->bufsize = b->maxsize;
return sndbuf_resize(b, 2, b->maxsize / 2);
}
void
sndbuf_free(struct snd_dbuf *b)
{
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);
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);
}
b->tmpbuf = NULL;
b->shadbuf = NULL;
b->buf = NULL;
b->sl = 0;
b->dmatag = NULL;
b->dmamap = NULL;
}
#define SNDBUF_CACHE_SHIFT 5
int
sndbuf_resize(struct snd_dbuf *b, unsigned int blkcnt, unsigned int blksz)
{
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
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_LOCK(b->channel);
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;
if (blkcnt == 0)
blkcnt = b->blkcnt;
if (blksz == 0)
blksz = b->blksz;
if (blkcnt < 2 || blksz < 16 || (blkcnt * blksz) > b->maxsize) {
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_UNLOCK(b->channel);
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
}
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;
2003-04-20 17:08:56 +00:00
bufsize = blkcnt * blksz;
Last (again ?!?) major commit for RELENG_7, featuring total Giant eradication in/from userland path, countless locking fixes, etc. - General sleep call through msleep(9) has been converted to condvar(9) with better consistencies. - Heavily guard every possible "slow path" entries (open(), close(), few ioctl()s, sysctls), but once it entering "fast path" (io, interrupt started), they are free to fly on their own. - Rearrange locking sequences, resulting better concurrency and serialization. Large part doesn't even need locking at all, and will be removed in future. Less clutter, except in few places due to lock ordering. - Anonymous mixer object creation/deletion to simplify mixer handling beyond typical mixer ioctls. Submitted by: chibis (with modifications) - Add few mix_[get|set|..] functions to avoid calling mixer_ioctl() directly using cryptic arguments. - Locking fixes to avoid possible deadlock with (still under Giant) USB. - Better simplex/duplex device handling. - Recover mmap() functionality for recording, which has been lost since 2.2.x - 3.x (the introduction of newpcm). Full-duplex mmap still doesn't work (due to VM/page design), but people still can mmap both by opening each direction separately. mmaped playback is guarantee to work either way. - New sysctl: "hw.snd.compat_linux_mmap" to allow PROT_EXEC page mapping, due to recent changes in linux compatibility layer which require it. All linux applications that using sound + mmap() (mostly games) require this to be enabled. Disabled by default. - Other goodies.. too many, that will increase releng7 shareholder value and make users of releng6 (and below) cry ;) * This commit should be atomic. If anything goes wrong (not counting problem originated from elsewhere), I will not hesitate to revert everything back within 12 hours. This substantial changes itself not a rocket science and the process has begun for almost 2 years, and lots of incremental changes are already in place during that period of time. * Some issues does occur in snd_emu10kx (note the 'x') due to various internal locking issues and it is currently being worked on by chibis. Tested by: chibis (Yuriy Tsibizov), joel, Alexandre Vieira, many innocent souls...
2007-06-16 03:37:28 +00:00
if (bufsize > b->allocsize ||
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
allocsize = round_page(bufsize);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_UNLOCK(b->channel);
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_LOCK(b->channel);
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;
b->bufsize = bufsize;
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:
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_UNLOCK(b->channel);
return 0;
}
int
sndbuf_remalloc(struct snd_dbuf *b, unsigned int blkcnt, unsigned int blksz)
{
unsigned int bufsize, allocsize;
u_int8_t *buf, *tmpbuf, *shadbuf;
if (blkcnt < 2 || blksz < 16)
return EINVAL;
bufsize = blksz * blkcnt;
if (bufsize > b->allocsize ||
bufsize < (b->allocsize >> SNDBUF_CACHE_SHIFT)) {
allocsize = round_page(bufsize);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_UNLOCK(b->channel);
buf = malloc(allocsize, M_DEVBUF, M_WAITOK);
tmpbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
shadbuf = malloc(allocsize, M_DEVBUF, M_WAITOK);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
CHN_LOCK(b->channel);
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);
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;
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);
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 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
*/
void
sndbuf_clear(struct snd_dbuf *b, unsigned int length)
{
int i;
2001-03-05 16:45:38 +00:00
u_char data, *p;
if (length == 0)
return;
2001-03-05 16:45:38 +00:00
if (length > b->bufsize)
length = b->bufsize;
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);
i = sndbuf_getfreeptr(b);
p = sndbuf_getbuf(b);
2001-03-05 16:45:38 +00:00
while (length > 0) {
p[i] = data;
length--;
i++;
if (i >= b->bufsize)
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
*/
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);
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);
}
void
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;
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);
}
u_int32_t
sndbuf_getfmt(struct snd_dbuf *b)
{
return b->fmt;
}
int
sndbuf_setfmt(struct snd_dbuf *b, u_int32_t fmt)
{
b->fmt = fmt;
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
b->bps = AFMT_BPS(b->fmt);
b->align = AFMT_ALIGN(b->fmt);
#if 0
b->bps = AFMT_CHANNEL(b->fmt);
Whats New: 1. Support wide range sampling rate, as low as 1hz up to int32 max (which is, insane) through new feeder_rate, multiple precisions choice (32/64 bit converter). This is indeed, quite insane, but it does give us more room and flexibility. Plenty sysctl options to adjust resampling characteristics. 2. Support 24/32 bit pcm format conversion through new, much improved, simplified and optimized feeder_fmt. Changes: 1. buffer.c / dsp.c / sound.h * Support for 24/32 AFMT. 2. feeder_rate.c * New implementation of sampling rate conversion with 32/64 bit precision, 1 - int32max hz (which is, ridiculous, yet very addictive). Much improved / smarter buffer management to not cause any missing samples at the end of conversion process * Tunable sysctls for various aspect: hw.snd.feeder_rate_ratemin - minimum allowable sampling rate (default to 4000) hw.snd.feeder_rate_ratemax - maximum allowable sampling rate (default to 1102500) hw.snd.feeder_rate_buffersize - conversion buffer size (default to 8192) hw.snd.feeder_rate_scaling - scaling / conversion method (please refer to the source for explaination). Default to previous implementation type. 3. feeder_fmt.c / sound.h * New implementation, support for 24/32bit conversion, optimized, and simplified. Few routines has been removed (8 to xlaw, 16 to 8). It just doesn't make sense. 4. channel.c * Support for 24/32 AFMT * Fix wrong xruns increment, causing incorrect underruns statistic while using vchans. 5. vchan.c * Support for 24/32 AFMT * Proper speed / rate detection especially for fixed rate ac97. User can override it using kernel hint: hint.pcm.<unit>.vchanrate="xxxx". Notes / Issues: * Virtual Channels (vchans) Enabling vchans can really, really help to solve overrun issues. This is quite understandable, because it operates entirely within its own buffering system without relying on hardware interrupt / state. Even if you don't need vchan, just enable single channel can help much. Few soundcards (notably via8233x, sblive, possibly others) have their own hardware multi channel, and this is unfortunately beyond vchan reachability. * The arrival of 24/32 also come with a price. Applications that can do 24/32bit playback need to be recompiled (notably mplayer). Use (recompiled) mplayer to experiment / test / debug this various format using -af format=fmt. Note that 24bit seeking in mplayer is a little bit broken, sometimes can cause silence or loud static noise. Pausing / seeking few times can solve this problem. You don't have to rebuild world entirely for this. Simply copy /usr/src/sys/sys/soundcard.h to /usr/include/sys/soundcard.h would suffice. Few drivers also need recompilation, and this can be done via /usr/src/sys/modules/sound/. Support for 24bit hardware playback is beyond the scope of this changes. That would require spessific hardware driver changes. * Don't expect playing 9999999999hz is a wise decision. Be reasonable. The new feeder_rate implemention provide flexibility, not insanity. You can easily chew up your CPU with this kind of mind instability. Please use proper mosquito repellent device for this obvious cracked brain attempt. As for testing purposes, you can use (again) mplayer to generate / play with different sampling rate. Use something like "mplayer -af resample=192000:0:0 <files>". Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my> Tested by: multimedia@
2005-07-31 16:16:22 +00:00
if (b->fmt & AFMT_16BIT)
b->bps <<= 1;
else if (b->fmt & AFMT_24BIT)
b->bps *= 3;
else if (b->fmt & AFMT_32BIT)
b->bps <<= 2;
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#endif
return 0;
}
unsigned int
sndbuf_getspd(struct snd_dbuf *b)
{
return b->spd;
}
void
sndbuf_setspd(struct snd_dbuf *b, unsigned int spd)
{
b->spd = spd;
}
unsigned int
sndbuf_getalign(struct snd_dbuf *b)
{
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
return (b->align);
}
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)
{
return b->bps;
}
void *
sndbuf_getbuf(struct snd_dbuf *b)
{
return b->buf;
}
void *
sndbuf_getbufofs(struct snd_dbuf *b, unsigned int ofs)
{
KASSERT(ofs < b->bufsize, ("%s: ofs invalid %d", __func__, ofs));
return b->buf + ofs;
}
unsigned int
sndbuf_getsize(struct snd_dbuf *b)
{
return b->bufsize;
}
unsigned int
sndbuf_getmaxsize(struct snd_dbuf *b)
{
return b->maxsize;
}
unsigned int
sndbuf_getallocsize(struct snd_dbuf *b)
{
return b->allocsize;
}
unsigned int
sndbuf_runsz(struct snd_dbuf *b)
{
return b->dl;
}
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
sndbuf_setxrun(struct snd_dbuf *b, unsigned int xrun)
{
SNDBUF_LOCKASSERT(b);
b->xrun = xrun;
}
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);
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
return b->rl;
}
unsigned int
sndbuf_getreadyptr(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
KASSERT((b->rp >= 0) && (b->rp <= b->bufsize), ("%s: b->rp invalid %d", __func__, b->rp));
return b->rp;
}
unsigned int
sndbuf_getfree(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d", __func__, b->rl));
return b->bufsize - b->rl;
}
unsigned int
sndbuf_getfreeptr(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
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));
return (b->rp + b->rl) % b->bufsize;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
u_int64_t
sndbuf_getblocks(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->total / b->blksz;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
u_int64_t
sndbuf_getprevblocks(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->prev_total / b->blksz;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
u_int64_t
sndbuf_gettotal(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->total;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
u_int64_t
sndbuf_getprevtotal(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
return b->prev_total;
}
void
sndbuf_updateprevtotal(struct snd_dbuf *b)
{
SNDBUF_LOCKASSERT(b);
b->prev_total = b->total;
}
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman <oberman@es.net> * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp> Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
unsigned int
sndbuf_xbytes(unsigned int v, struct snd_dbuf *from, struct snd_dbuf *to)
{
if (from == NULL || to == NULL || v == 0)
return 0;
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
return snd_xbytes(v, sndbuf_getalign(from) * sndbuf_getspd(from),
sndbuf_getalign(to) * sndbuf_getspd(to));
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman <oberman@es.net> * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp> Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
}
u_int8_t
sndbuf_zerodata(u_int32_t fmt)
{
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
if (fmt & (AFMT_SIGNED | AFMT_PASSTHROUGH))
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman <oberman@es.net> * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp> Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
return (0x00);
else if (fmt & AFMT_MU_LAW)
return (0x7f);
else if (fmt & AFMT_A_LAW)
return (0x55);
return (0x80);
}
/************************************************************/
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
*/
int
sndbuf_acquire(struct snd_dbuf *b, u_int8_t *from, unsigned int count)
{
int l;
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));
b->total += count;
if (from != NULL) {
while (count > 0) {
l = min(count, sndbuf_getsize(b) - sndbuf_getfreeptr(b));
bcopy(from, sndbuf_getbufofs(b, sndbuf_getfreeptr(b)), l);
from += l;
b->rl += l;
count -= l;
}
} else
b->rl += count;
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d, count %d", __func__, b->rl, count));
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
*/
int
sndbuf_dispose(struct snd_dbuf *b, u_int8_t *to, unsigned int count)
{
int l;
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));
if (to != NULL) {
while (count > 0) {
l = min(count, sndbuf_getsize(b) - sndbuf_getreadyptr(b));
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;
}
KASSERT((b->rl >= 0) && (b->rl <= b->bufsize), ("%s: b->rl invalid %d, count %d", __func__, b->rl, count));
return 0;
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#ifdef SND_DIAGNOSTIC
static uint32_t snd_feeder_maxfeed = 0;
SYSCTL_UINT(_hw_snd, OID_AUTO, feeder_maxfeed, CTLFLAG_RD,
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
&snd_feeder_maxfeed, 0, "maximum feeder count request");
static uint32_t snd_feeder_maxcycle = 0;
SYSCTL_UINT(_hw_snd, OID_AUTO, feeder_maxcycle, CTLFLAG_RD,
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
&snd_feeder_maxcycle, 0, "maximum feeder cycle");
#endif
/* count is number of bytes we want added to destination buffer */
int
sndbuf_feed(struct snd_dbuf *from, struct snd_dbuf *to, struct pcm_channel *channel, struct pcm_feeder *feeder, unsigned int count)
{
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
unsigned int cnt, maxfeed;
#ifdef SND_DIAGNOSTIC
unsigned int cycle;
if (count > snd_feeder_maxfeed)
snd_feeder_maxfeed = count;
cycle = 0;
#endif
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman <oberman@es.net> * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp> Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
KASSERT(count > 0, ("can't feed 0 bytes"));
if (sndbuf_getfree(to) < count)
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
return (EINVAL);
maxfeed = SND_FXROUND(SND_FXDIV_MAX, sndbuf_getalign(to));
Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes in every sense. General ------- - Multichannel safe, endian safe, format safe * Large part of critical pcm filters such as vchan.c, feeder_rate.c, feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that using them does not cause the pcm data to be converted to 16bit little endian. * Macrosses for accessing pcm data safely are defined within sound.h in the form of PCM_READ_* / PCM_WRITE_* * Currently, most of them are probably limited for mono/stereo handling, but the future addition of true multichannel will be much easier. - Low latency operation * Well, this require lot more works to do not just within sound driver, but we're heading towards right direction. Buffer/block sizing within channel.c is rewritten to calculate precise allocation for various combination of sample/data/rate size. As a result, applying correct SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar to what commercial 4front driver do. * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not result long delay. * Eliminate sound truncation if the sound data is too small. DIY: 1) Download / extract http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz 2) Do a comparison between "cat state*.au > /dev/dsp" and "for x in state*.au ; do cat $x > /dev/dsp ; done" - there should be no "perceivable" differences. Double close for PR kern/31445. CAVEAT: Low latency come with (unbearable) price especially for poorly written applications. Applications that trying to act smarter by requesting (wrong) blocksize/blockcount will suffer the most. Fixup samples/patches can be found at: http://people.freebsd.org/~ariff/ports/ - Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42) due to closer compatibility with 4front driver. Discussed with: marcus@ (long time ago?) - All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been moved to their own dev sysctl nodes, notably: hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans Bump __FreeBSD_version. Driver specific --------------- - Ditto for sysctls. - snd_atiixp, snd_es137x, snd_via8233, snd_hda * Numerous cleanups and fixes. * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme. This was intended for pure debugging and latency measurement, but proven good enough in few unexpected and rare cases (such as problematic shared IRQ with GIANT devices - USB). Polling can be enabled/disabled through dev.pcm.0.polling. Disabled by default. - snd_ich * Fix possible overflow during speed calibration. Delay final initialization (pcm_setstatus) after calibration finished. PR: kern/100169 Tested by: Kevin Overman <oberman@es.net> * Inverted EAPD for few Nec VersaPro. PR: kern/104715 Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp> Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman, those at #freebsd-azalia @ freenode and others for testing. Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
do {
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
cnt = FEEDER_FEED(feeder, channel, to->tmpbuf,
min(count, maxfeed), from);
if (cnt == 0)
break;
sndbuf_acquire(to, to->tmpbuf, cnt);
count -= cnt;
#ifdef SND_DIAGNOSTIC
cycle++;
#endif
} while (count != 0);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#ifdef SND_DIAGNOSTIC
if (cycle > snd_feeder_maxcycle)
snd_feeder_maxcycle = cycle;
#endif
return (0);
}
/************************************************************/
void
sndbuf_dump(struct snd_dbuf *b, char *s, u_int32_t what)
{
printf("%s: [", s);
if (what & 0x01)
printf(" bufsize: %d, maxsize: %d", b->bufsize, b->maxsize);
if (what & 0x02)
printf(" dl: %d, rp: %d, rl: %d, hp: %d", b->dl, b->rp, b->rl, b->hp);
if (what & 0x04)
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
printf(" total: %ju, prev_total: %ju, xrun: %d", (uintmax_t)b->total, (uintmax_t)b->prev_total, b->xrun);
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