freebsd-skq/sys/dev/sound/pci/csapcm.c

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/*-
* Copyright (c) 1999 Seigo Tanimura
* All rights reserved.
*
* Portions of this source are based on cwcealdr.cpp and dhwiface.cpp in
* cwcealdr1.zip, the sample sources by Crystal Semiconductor.
* Copyright (c) 1996-1998 Crystal Semiconductor Corp.
*
* 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 <dev/sound/pcm/ac97.h>
#include <dev/sound/chip.h>
#include <dev/sound/pci/csareg.h>
#include <dev/sound/pci/csavar.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
SND_DECLARE_FILE("$FreeBSD$");
/* Buffer size on dma transfer. Fixed for CS416x. */
#define CS461x_BUFFSIZE (4 * 1024)
#define GOF_PER_SEC 200
/* device private data */
struct csa_info;
struct csa_chinfo {
struct csa_info *parent;
struct pcm_channel *channel;
struct snd_dbuf *buffer;
int dir;
u_int32_t fmt, spd;
int dma;
};
struct csa_info {
csa_res res; /* resource */
void *ih; /* Interrupt cookie */
bus_dma_tag_t parent_dmat; /* DMA tag */
struct csa_bridgeinfo *binfo; /* The state of the parent. */
struct csa_card *card;
int active;
/* Contents of board's registers */
u_long pfie;
u_long pctl;
u_long cctl;
struct csa_chinfo pch, rch;
u_int32_t ac97[CS461x_AC97_NUMBER_RESTORE_REGS];
u_int32_t ac97_powerdown;
u_int32_t ac97_general_purpose;
};
/* -------------------------------------------------------------------- */
/* prototypes */
static int csa_init(struct csa_info *);
static void csa_intr(void *);
static void csa_setplaysamplerate(csa_res *resp, u_long ulInRate);
static void csa_setcapturesamplerate(csa_res *resp, u_long ulOutRate);
static void csa_startplaydma(struct csa_info *csa);
static void csa_startcapturedma(struct csa_info *csa);
static void csa_stopplaydma(struct csa_info *csa);
static void csa_stopcapturedma(struct csa_info *csa);
static int csa_startdsp(csa_res *resp);
static int csa_stopdsp(csa_res *resp);
static int csa_allocres(struct csa_info *scp, device_t dev);
static void csa_releaseres(struct csa_info *scp, device_t dev);
static void csa_ac97_suspend(struct csa_info *csa);
static void csa_ac97_resume(struct csa_info *csa);
static u_int32_t csa_playfmt[] = {
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_FORMAT(AFMT_U8, 1, 0),
SND_FORMAT(AFMT_U8, 2, 0),
SND_FORMAT(AFMT_S8, 1, 0),
SND_FORMAT(AFMT_S8, 2, 0),
SND_FORMAT(AFMT_S16_LE, 1, 0),
SND_FORMAT(AFMT_S16_LE, 2, 0),
SND_FORMAT(AFMT_S16_BE, 1, 0),
SND_FORMAT(AFMT_S16_BE, 2, 0),
0
};
static struct pcmchan_caps csa_playcaps = {8000, 48000, csa_playfmt, 0};
static u_int32_t csa_recfmt[] = {
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_FORMAT(AFMT_S16_LE, 1, 0),
SND_FORMAT(AFMT_S16_LE, 2, 0),
0
};
static struct pcmchan_caps csa_reccaps = {11025, 48000, csa_recfmt, 0};
/* -------------------------------------------------------------------- */
static int
csa_active(struct csa_info *csa, int run)
{
int old;
old = csa->active;
csa->active += run;
if ((csa->active > 1) || (csa->active < -1))
csa->active = 0;
if (csa->card->active)
return (csa->card->active(!(csa->active && old)));
return 0;
}
/* -------------------------------------------------------------------- */
/* ac97 codec */
static int
csa_rdcd(kobj_t obj, void *devinfo, int regno)
{
u_int32_t data;
struct csa_info *csa = (struct csa_info *)devinfo;
csa_active(csa, 1);
if (csa_readcodec(&csa->res, regno + BA0_AC97_RESET, &data))
data = 0;
csa_active(csa, -1);
return data;
}
static int
csa_wrcd(kobj_t obj, void *devinfo, int regno, u_int32_t data)
{
struct csa_info *csa = (struct csa_info *)devinfo;
csa_active(csa, 1);
csa_writecodec(&csa->res, regno + BA0_AC97_RESET, data);
csa_active(csa, -1);
return 0;
}
static kobj_method_t csa_ac97_methods[] = {
KOBJMETHOD(ac97_read, csa_rdcd),
KOBJMETHOD(ac97_write, csa_wrcd),
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
KOBJMETHOD_END
};
AC97_DECLARE(csa_ac97);
static void
csa_setplaysamplerate(csa_res *resp, u_long ulInRate)
{
u_long ulTemp1, ulTemp2;
u_long ulPhiIncr;
u_long ulCorrectionPerGOF, ulCorrectionPerSec;
u_long ulOutRate;
ulOutRate = 48000;
/*
* Compute the values used to drive the actual sample rate conversion.
* The following formulas are being computed, using inline assembly
* since we need to use 64 bit arithmetic to compute the values:
*
* ulPhiIncr = floor((Fs,in * 2^26) / Fs,out)
* ulCorrectionPerGOF = floor((Fs,in * 2^26 - Fs,out * ulPhiIncr) /
* GOF_PER_SEC)
* ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -
* GOF_PER_SEC * ulCorrectionPerGOF
*
* i.e.
*
* ulPhiIncr:ulOther = dividend:remainder((Fs,in * 2^26) / Fs,out)
* ulCorrectionPerGOF:ulCorrectionPerSec =
* dividend:remainder(ulOther / GOF_PER_SEC)
*/
ulTemp1 = ulInRate << 16;
ulPhiIncr = ulTemp1 / ulOutRate;
ulTemp1 -= ulPhiIncr * ulOutRate;
ulTemp1 <<= 10;
ulPhiIncr <<= 10;
ulTemp2 = ulTemp1 / ulOutRate;
ulPhiIncr += ulTemp2;
ulTemp1 -= ulTemp2 * ulOutRate;
ulCorrectionPerGOF = ulTemp1 / GOF_PER_SEC;
ulTemp1 -= ulCorrectionPerGOF * GOF_PER_SEC;
ulCorrectionPerSec = ulTemp1;
/*
* Fill in the SampleRateConverter control block.
*/
csa_writemem(resp, BA1_PSRC, ((ulCorrectionPerSec << 16) & 0xFFFF0000) | (ulCorrectionPerGOF & 0xFFFF));
csa_writemem(resp, BA1_PPI, ulPhiIncr);
}
static void
csa_setcapturesamplerate(csa_res *resp, u_long ulOutRate)
{
u_long ulPhiIncr, ulCoeffIncr, ulTemp1, ulTemp2;
u_long ulCorrectionPerGOF, ulCorrectionPerSec, ulInitialDelay;
u_long dwFrameGroupLength, dwCnt;
u_long ulInRate;
ulInRate = 48000;
/*
* We can only decimate by up to a factor of 1/9th the hardware rate.
* Return an error if an attempt is made to stray outside that limit.
*/
if((ulOutRate * 9) < ulInRate)
return;
/*
* We can not capture at at rate greater than the Input Rate (48000).
* Return an error if an attempt is made to stray outside that limit.
*/
if(ulOutRate > ulInRate)
return;
/*
* Compute the values used to drive the actual sample rate conversion.
* The following formulas are being computed, using inline assembly
* since we need to use 64 bit arithmetic to compute the values:
*
* ulCoeffIncr = -floor((Fs,out * 2^23) / Fs,in)
* ulPhiIncr = floor((Fs,in * 2^26) / Fs,out)
* ulCorrectionPerGOF = floor((Fs,in * 2^26 - Fs,out * ulPhiIncr) /
* GOF_PER_SEC)
* ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -
* GOF_PER_SEC * ulCorrectionPerGOF
* ulInitialDelay = ceil((24 * Fs,in) / Fs,out)
*
* i.e.
*
* ulCoeffIncr = neg(dividend((Fs,out * 2^23) / Fs,in))
* ulPhiIncr:ulOther = dividend:remainder((Fs,in * 2^26) / Fs,out)
* ulCorrectionPerGOF:ulCorrectionPerSec =
* dividend:remainder(ulOther / GOF_PER_SEC)
* ulInitialDelay = dividend(((24 * Fs,in) + Fs,out - 1) / Fs,out)
*/
ulTemp1 = ulOutRate << 16;
ulCoeffIncr = ulTemp1 / ulInRate;
ulTemp1 -= ulCoeffIncr * ulInRate;
ulTemp1 <<= 7;
ulCoeffIncr <<= 7;
ulCoeffIncr += ulTemp1 / ulInRate;
ulCoeffIncr ^= 0xFFFFFFFF;
ulCoeffIncr++;
ulTemp1 = ulInRate << 16;
ulPhiIncr = ulTemp1 / ulOutRate;
ulTemp1 -= ulPhiIncr * ulOutRate;
ulTemp1 <<= 10;
ulPhiIncr <<= 10;
ulTemp2 = ulTemp1 / ulOutRate;
ulPhiIncr += ulTemp2;
ulTemp1 -= ulTemp2 * ulOutRate;
ulCorrectionPerGOF = ulTemp1 / GOF_PER_SEC;
ulTemp1 -= ulCorrectionPerGOF * GOF_PER_SEC;
ulCorrectionPerSec = ulTemp1;
ulInitialDelay = ((ulInRate * 24) + ulOutRate - 1) / ulOutRate;
/*
* Fill in the VariDecimate control block.
*/
csa_writemem(resp, BA1_CSRC,
((ulCorrectionPerSec << 16) & 0xFFFF0000) | (ulCorrectionPerGOF & 0xFFFF));
csa_writemem(resp, BA1_CCI, ulCoeffIncr);
csa_writemem(resp, BA1_CD,
(((BA1_VARIDEC_BUF_1 + (ulInitialDelay << 2)) << 16) & 0xFFFF0000) | 0x80);
csa_writemem(resp, BA1_CPI, ulPhiIncr);
/*
* Figure out the frame group length for the write back task. Basically,
* this is just the factors of 24000 (2^6*3*5^3) that are not present in
* the output sample rate.
*/
dwFrameGroupLength = 1;
for(dwCnt = 2; dwCnt <= 64; dwCnt *= 2)
{
if(((ulOutRate / dwCnt) * dwCnt) !=
ulOutRate)
{
dwFrameGroupLength *= 2;
}
}
if(((ulOutRate / 3) * 3) !=
ulOutRate)
{
dwFrameGroupLength *= 3;
}
for(dwCnt = 5; dwCnt <= 125; dwCnt *= 5)
{
if(((ulOutRate / dwCnt) * dwCnt) !=
ulOutRate)
{
dwFrameGroupLength *= 5;
}
}
/*
* Fill in the WriteBack control block.
*/
csa_writemem(resp, BA1_CFG1, dwFrameGroupLength);
csa_writemem(resp, BA1_CFG2, (0x00800000 | dwFrameGroupLength));
csa_writemem(resp, BA1_CCST, 0x0000FFFF);
csa_writemem(resp, BA1_CSPB, ((65536 * ulOutRate) / 24000));
csa_writemem(resp, (BA1_CSPB + 4), 0x0000FFFF);
}
static void
csa_startplaydma(struct csa_info *csa)
{
csa_res *resp;
u_long ul;
if (!csa->pch.dma) {
resp = &csa->res;
ul = csa_readmem(resp, BA1_PCTL);
ul &= 0x0000ffff;
csa_writemem(resp, BA1_PCTL, ul | csa->pctl);
csa_writemem(resp, BA1_PVOL, 0x80008000);
csa->pch.dma = 1;
}
}
static void
csa_startcapturedma(struct csa_info *csa)
{
csa_res *resp;
u_long ul;
if (!csa->rch.dma) {
resp = &csa->res;
ul = csa_readmem(resp, BA1_CCTL);
ul &= 0xffff0000;
csa_writemem(resp, BA1_CCTL, ul | csa->cctl);
csa_writemem(resp, BA1_CVOL, 0x80008000);
csa->rch.dma = 1;
}
}
static void
csa_stopplaydma(struct csa_info *csa)
{
csa_res *resp;
u_long ul;
if (csa->pch.dma) {
resp = &csa->res;
ul = csa_readmem(resp, BA1_PCTL);
csa->pctl = ul & 0xffff0000;
csa_writemem(resp, BA1_PCTL, ul & 0x0000ffff);
csa_writemem(resp, BA1_PVOL, 0xffffffff);
csa->pch.dma = 0;
/*
* The bitwise pointer of the serial FIFO in the DSP
* seems to make an error upon starting or stopping the
* DSP. Clear the FIFO and correct the pointer if we
* are not capturing.
*/
if (!csa->rch.dma) {
csa_clearserialfifos(resp);
csa_writeio(resp, BA0_SERBSP, 0);
}
}
}
static void
csa_stopcapturedma(struct csa_info *csa)
{
csa_res *resp;
u_long ul;
if (csa->rch.dma) {
resp = &csa->res;
ul = csa_readmem(resp, BA1_CCTL);
csa->cctl = ul & 0x0000ffff;
csa_writemem(resp, BA1_CCTL, ul & 0xffff0000);
csa_writemem(resp, BA1_CVOL, 0xffffffff);
csa->rch.dma = 0;
/*
* The bitwise pointer of the serial FIFO in the DSP
* seems to make an error upon starting or stopping the
* DSP. Clear the FIFO and correct the pointer if we
* are not playing.
*/
if (!csa->pch.dma) {
csa_clearserialfifos(resp);
csa_writeio(resp, BA0_SERBSP, 0);
}
}
}
static int
csa_startdsp(csa_res *resp)
{
int i;
u_long ul;
/*
* Set the frame timer to reflect the number of cycles per frame.
*/
csa_writemem(resp, BA1_FRMT, 0xadf);
/*
* Turn on the run, run at frame, and DMA enable bits in the local copy of
* the SP control register.
*/
csa_writemem(resp, BA1_SPCR, SPCR_RUN | SPCR_RUNFR | SPCR_DRQEN);
/*
* Wait until the run at frame bit resets itself in the SP control
* register.
*/
ul = 0;
for (i = 0 ; i < 25 ; i++) {
/*
* Wait a little bit, so we don't issue PCI reads too frequently.
*/
DELAY(50);
/*
* Fetch the current value of the SP status register.
*/
ul = csa_readmem(resp, BA1_SPCR);
/*
* If the run at frame bit has reset, then stop waiting.
*/
if((ul & SPCR_RUNFR) == 0)
break;
}
/*
* If the run at frame bit never reset, then return an error.
*/
if((ul & SPCR_RUNFR) != 0)
return (EAGAIN);
return (0);
}
static int
csa_stopdsp(csa_res *resp)
{
/*
* Turn off the run, run at frame, and DMA enable bits in
* the local copy of the SP control register.
*/
csa_writemem(resp, BA1_SPCR, 0);
return (0);
}
static int
csa_setupchan(struct csa_chinfo *ch)
{
struct csa_info *csa = ch->parent;
csa_res *resp = &csa->res;
u_long pdtc, tmp;
if (ch->dir == PCMDIR_PLAY) {
/* direction */
csa_writemem(resp, BA1_PBA, sndbuf_getbufaddr(ch->buffer));
/* format */
csa->pfie = csa_readmem(resp, BA1_PFIE) & ~0x0000f03f;
if (!(ch->fmt & AFMT_SIGNED))
csa->pfie |= 0x8000;
if (ch->fmt & AFMT_BIGENDIAN)
csa->pfie |= 0x4000;
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 (AFMT_CHANNEL(ch->fmt) < 2)
csa->pfie |= 0x2000;
if (ch->fmt & AFMT_8BIT)
csa->pfie |= 0x1000;
csa_writemem(resp, BA1_PFIE, csa->pfie);
tmp = 4;
if (ch->fmt & AFMT_16BIT)
tmp <<= 1;
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 (AFMT_CHANNEL(ch->fmt) > 1)
tmp <<= 1;
tmp--;
pdtc = csa_readmem(resp, BA1_PDTC) & ~0x000001ff;
pdtc |= tmp;
csa_writemem(resp, BA1_PDTC, pdtc);
/* rate */
csa_setplaysamplerate(resp, ch->spd);
} else if (ch->dir == PCMDIR_REC) {
/* direction */
csa_writemem(resp, BA1_CBA, sndbuf_getbufaddr(ch->buffer));
/* format */
csa_writemem(resp, BA1_CIE, (csa_readmem(resp, BA1_CIE) & ~0x0000003f) | 0x00000001);
/* rate */
csa_setcapturesamplerate(resp, ch->spd);
}
return 0;
}
/* -------------------------------------------------------------------- */
/* channel interface */
static void *
csachan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
{
struct csa_info *csa = devinfo;
struct csa_chinfo *ch = (dir == PCMDIR_PLAY)? &csa->pch : &csa->rch;
ch->parent = csa;
ch->channel = c;
ch->buffer = b;
ch->dir = dir;
if (sndbuf_alloc(ch->buffer, csa->parent_dmat, 0, CS461x_BUFFSIZE) != 0)
return NULL;
return ch;
}
static int
csachan_setformat(kobj_t obj, void *data, u_int32_t format)
{
struct csa_chinfo *ch = data;
ch->fmt = format;
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
static u_int32_t
csachan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct csa_chinfo *ch = data;
ch->spd = speed;
return ch->spd; /* XXX calc real speed */
}
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
static u_int32_t
csachan_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
return CS461x_BUFFSIZE / 2;
}
static int
csachan_trigger(kobj_t obj, void *data, int go)
{
struct csa_chinfo *ch = data;
struct csa_info *csa = ch->parent;
if (!PCMTRIG_COMMON(go))
return 0;
if (go == PCMTRIG_START) {
csa_active(csa, 1);
csa_setupchan(ch);
if (ch->dir == PCMDIR_PLAY)
csa_startplaydma(csa);
else
csa_startcapturedma(csa);
} else {
if (ch->dir == PCMDIR_PLAY)
csa_stopplaydma(csa);
else
csa_stopcapturedma(csa);
csa_active(csa, -1);
}
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
static u_int32_t
csachan_getptr(kobj_t obj, void *data)
{
struct csa_chinfo *ch = data;
struct csa_info *csa = ch->parent;
csa_res *resp;
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_int32_t ptr;
resp = &csa->res;
if (ch->dir == PCMDIR_PLAY) {
ptr = csa_readmem(resp, BA1_PBA) - sndbuf_getbufaddr(ch->buffer);
if ((ch->fmt & AFMT_U8) != 0 || (ch->fmt & AFMT_S8) != 0)
ptr >>= 1;
} else {
ptr = csa_readmem(resp, BA1_CBA) - sndbuf_getbufaddr(ch->buffer);
if ((ch->fmt & AFMT_U8) != 0 || (ch->fmt & AFMT_S8) != 0)
ptr >>= 1;
}
return (ptr);
}
static struct pcmchan_caps *
csachan_getcaps(kobj_t obj, void *data)
{
struct csa_chinfo *ch = data;
return (ch->dir == PCMDIR_PLAY)? &csa_playcaps : &csa_reccaps;
}
static kobj_method_t csachan_methods[] = {
KOBJMETHOD(channel_init, csachan_init),
KOBJMETHOD(channel_setformat, csachan_setformat),
KOBJMETHOD(channel_setspeed, csachan_setspeed),
KOBJMETHOD(channel_setblocksize, csachan_setblocksize),
KOBJMETHOD(channel_trigger, csachan_trigger),
KOBJMETHOD(channel_getptr, csachan_getptr),
KOBJMETHOD(channel_getcaps, csachan_getcaps),
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
KOBJMETHOD_END
};
CHANNEL_DECLARE(csachan);
/* -------------------------------------------------------------------- */
/* The interrupt handler */
static void
csa_intr(void *p)
{
struct csa_info *csa = p;
if ((csa->binfo->hisr & HISR_VC0) != 0)
chn_intr(csa->pch.channel);
if ((csa->binfo->hisr & HISR_VC1) != 0)
chn_intr(csa->rch.channel);
}
/* -------------------------------------------------------------------- */
/*
* Probe and attach the card
*/
static int
csa_init(struct csa_info *csa)
{
csa_res *resp;
resp = &csa->res;
csa->pfie = 0;
csa_stopplaydma(csa);
csa_stopcapturedma(csa);
if (csa_startdsp(resp))
return (1);
/* Crank up the power on the DAC and ADC. */
csa_setplaysamplerate(resp, 8000);
csa_setcapturesamplerate(resp, 8000);
/* Set defaults */
csa_writeio(resp, BA0_EGPIODR, EGPIODR_GPOE0);
csa_writeio(resp, BA0_EGPIOPTR, EGPIOPTR_GPPT0);
/* Power up amplifier */
csa_writeio(resp, BA0_EGPIODR, csa_readio(resp, BA0_EGPIODR) |
EGPIODR_GPOE2);
csa_writeio(resp, BA0_EGPIOPTR, csa_readio(resp, BA0_EGPIOPTR) |
EGPIOPTR_GPPT2);
return 0;
}
/* Allocates resources. */
static int
csa_allocres(struct csa_info *csa, device_t dev)
{
csa_res *resp;
resp = &csa->res;
if (resp->io == NULL) {
resp->io = bus_alloc_resource_any(dev, SYS_RES_MEMORY,
&resp->io_rid, RF_ACTIVE);
if (resp->io == NULL)
return (1);
}
if (resp->mem == NULL) {
resp->mem = bus_alloc_resource_any(dev, SYS_RES_MEMORY,
&resp->mem_rid, RF_ACTIVE);
if (resp->mem == NULL)
return (1);
}
if (resp->irq == NULL) {
resp->irq = bus_alloc_resource_any(dev, SYS_RES_IRQ,
&resp->irq_rid, RF_ACTIVE | RF_SHAREABLE);
if (resp->irq == NULL)
return (1);
}
if (bus_dma_tag_create(/*parent*/bus_get_dma_tag(dev),
/*alignment*/CS461x_BUFFSIZE,
/*boundary*/CS461x_BUFFSIZE,
/*lowaddr*/BUS_SPACE_MAXADDR_32BIT,
/*highaddr*/BUS_SPACE_MAXADDR,
/*filter*/NULL, /*filterarg*/NULL,
/*maxsize*/CS461x_BUFFSIZE, /*nsegments*/1, /*maxsegz*/0x3ffff,
/*flags*/0, /*lockfunc*/busdma_lock_mutex,
/*lockarg*/&Giant, &csa->parent_dmat) != 0)
return (1);
return (0);
}
/* Releases resources. */
static void
csa_releaseres(struct csa_info *csa, device_t dev)
{
csa_res *resp;
KASSERT(csa != NULL, ("called with bogus resource structure"));
resp = &csa->res;
if (resp->irq != NULL) {
if (csa->ih)
bus_teardown_intr(dev, resp->irq, csa->ih);
bus_release_resource(dev, SYS_RES_IRQ, resp->irq_rid, resp->irq);
resp->irq = NULL;
}
if (resp->io != NULL) {
bus_release_resource(dev, SYS_RES_MEMORY, resp->io_rid, resp->io);
resp->io = NULL;
}
if (resp->mem != NULL) {
bus_release_resource(dev, SYS_RES_MEMORY, resp->mem_rid, resp->mem);
resp->mem = NULL;
}
if (csa->parent_dmat != NULL) {
bus_dma_tag_destroy(csa->parent_dmat);
csa->parent_dmat = NULL;
}
free(csa, M_DEVBUF);
}
static int
pcmcsa_probe(device_t dev)
{
char *s;
struct sndcard_func *func;
/* The parent device has already been probed. */
func = device_get_ivars(dev);
if (func == NULL || func->func != SCF_PCM)
return (ENXIO);
s = "CS461x PCM Audio";
device_set_desc(dev, s);
return (0);
}
static int
pcmcsa_attach(device_t dev)
{
struct csa_info *csa;
csa_res *resp;
int unit;
char status[SND_STATUSLEN];
struct ac97_info *codec;
struct sndcard_func *func;
csa = malloc(sizeof(*csa), M_DEVBUF, M_WAITOK | M_ZERO);
unit = device_get_unit(dev);
func = device_get_ivars(dev);
csa->binfo = func->varinfo;
/*
* Fake the status of DMA so that the initial value of
* PCTL and CCTL can be stored into csa->pctl and csa->cctl,
* respectively.
*/
csa->pch.dma = csa->rch.dma = 1;
csa->active = 0;
csa->card = csa->binfo->card;
/* Allocate the resources. */
resp = &csa->res;
resp->io_rid = PCIR_BAR(0);
resp->mem_rid = PCIR_BAR(1);
resp->irq_rid = 0;
if (csa_allocres(csa, dev)) {
csa_releaseres(csa, dev);
return (ENXIO);
}
csa_active(csa, 1);
if (csa_init(csa)) {
csa_releaseres(csa, dev);
return (ENXIO);
}
codec = AC97_CREATE(dev, csa, csa_ac97);
if (codec == NULL) {
csa_releaseres(csa, dev);
return (ENXIO);
}
if (csa->card->inv_eapd)
ac97_setflags(codec, AC97_F_EAPD_INV);
if (mixer_init(dev, ac97_getmixerclass(), codec) == -1) {
ac97_destroy(codec);
csa_releaseres(csa, dev);
return (ENXIO);
}
snprintf(status, SND_STATUSLEN, "at irq %ld %s",
rman_get_start(resp->irq),PCM_KLDSTRING(snd_csa));
/* Enable interrupt. */
if (snd_setup_intr(dev, resp->irq, 0, csa_intr, csa, &csa->ih)) {
ac97_destroy(codec);
csa_releaseres(csa, dev);
return (ENXIO);
}
csa_writemem(resp, BA1_PFIE, csa_readmem(resp, BA1_PFIE) & ~0x0000f03f);
csa_writemem(resp, BA1_CIE, (csa_readmem(resp, BA1_CIE) & ~0x0000003f) | 0x00000001);
csa_active(csa, -1);
if (pcm_register(dev, csa, 1, 1)) {
ac97_destroy(codec);
csa_releaseres(csa, dev);
return (ENXIO);
}
pcm_addchan(dev, PCMDIR_REC, &csachan_class, csa);
pcm_addchan(dev, PCMDIR_PLAY, &csachan_class, csa);
pcm_setstatus(dev, status);
return (0);
}
static int
pcmcsa_detach(device_t dev)
{
int r;
struct csa_info *csa;
r = pcm_unregister(dev);
if (r)
return r;
csa = pcm_getdevinfo(dev);
csa_releaseres(csa, dev);
return 0;
}
static void
csa_ac97_suspend(struct csa_info *csa)
{
int count, i;
uint32_t tmp;
for (count = 0x2, i=0;
(count <= CS461x_AC97_HIGHESTREGTORESTORE) &&
(i < CS461x_AC97_NUMBER_RESTORE_REGS);
count += 2, i++)
csa_readcodec(&csa->res, BA0_AC97_RESET + count, &csa->ac97[i]);
/* mute the outputs */
csa_writecodec(&csa->res, BA0_AC97_MASTER_VOLUME, 0x8000);
csa_writecodec(&csa->res, BA0_AC97_HEADPHONE_VOLUME, 0x8000);
csa_writecodec(&csa->res, BA0_AC97_MASTER_VOLUME_MONO, 0x8000);
csa_writecodec(&csa->res, BA0_AC97_PCM_OUT_VOLUME, 0x8000);
/* save the registers that cause pops */
csa_readcodec(&csa->res, BA0_AC97_POWERDOWN, &csa->ac97_powerdown);
csa_readcodec(&csa->res, BA0_AC97_GENERAL_PURPOSE,
&csa->ac97_general_purpose);
/*
* And power down everything on the AC97 codec. Well, for now,
* only power down the DAC/ADC and MIXER VREFON components.
* trouble with removing VREF.
*/
/* MIXVON */
csa_readcodec(&csa->res, BA0_AC97_POWERDOWN, &tmp);
csa_writecodec(&csa->res, BA0_AC97_POWERDOWN,
tmp | CS_AC97_POWER_CONTROL_MIXVON);
/* ADC */
csa_readcodec(&csa->res, BA0_AC97_POWERDOWN, &tmp);
csa_writecodec(&csa->res, BA0_AC97_POWERDOWN,
tmp | CS_AC97_POWER_CONTROL_ADC);
/* DAC */
csa_readcodec(&csa->res, BA0_AC97_POWERDOWN, &tmp);
csa_writecodec(&csa->res, BA0_AC97_POWERDOWN,
tmp | CS_AC97_POWER_CONTROL_DAC);
}
static void
csa_ac97_resume(struct csa_info *csa)
{
int count, i;
/*
* First, we restore the state of the general purpose register. This
* contains the mic select (mic1 or mic2) and if we restore this after
* we restore the mic volume/boost state and mic2 was selected at
* suspend time, we will end up with a brief period of time where mic1
* is selected with the volume/boost settings for mic2, causing
* acoustic feedback. So we restore the general purpose register
* first, thereby getting the correct mic selected before we restore
* the mic volume/boost.
*/
csa_writecodec(&csa->res, BA0_AC97_GENERAL_PURPOSE,
csa->ac97_general_purpose);
/*
* Now, while the outputs are still muted, restore the state of power
* on the AC97 part.
*/
csa_writecodec(&csa->res, BA0_AC97_POWERDOWN, csa->ac97_powerdown);
/*
* Restore just the first set of registers, from register number
* 0x02 to the register number that ulHighestRegToRestore specifies.
*/
for (count = 0x2, i=0;
(count <= CS461x_AC97_HIGHESTREGTORESTORE) &&
(i < CS461x_AC97_NUMBER_RESTORE_REGS);
count += 2, i++)
csa_writecodec(&csa->res, BA0_AC97_RESET + count, csa->ac97[i]);
}
static int
pcmcsa_suspend(device_t dev)
{
struct csa_info *csa;
csa_res *resp;
csa = pcm_getdevinfo(dev);
resp = &csa->res;
csa_active(csa, 1);
/* playback interrupt disable */
csa_writemem(resp, BA1_PFIE,
(csa_readmem(resp, BA1_PFIE) & ~0x0000f03f) | 0x00000010);
/* capture interrupt disable */
csa_writemem(resp, BA1_CIE,
(csa_readmem(resp, BA1_CIE) & ~0x0000003f) | 0x00000011);
csa_stopplaydma(csa);
csa_stopcapturedma(csa);
csa_ac97_suspend(csa);
csa_resetdsp(resp);
csa_stopdsp(resp);
/*
* Power down the DAC and ADC. For now leave the other areas on.
*/
csa_writecodec(&csa->res, BA0_AC97_POWERDOWN, 0x300);
/*
* Power down the PLL.
*/
csa_writemem(resp, BA0_CLKCR1, 0);
/*
* Turn off the Processor by turning off the software clock
* enable flag in the clock control register.
*/
csa_writemem(resp, BA0_CLKCR1,
csa_readmem(resp, BA0_CLKCR1) & ~CLKCR1_SWCE);
csa_active(csa, -1);
return 0;
}
static int
pcmcsa_resume(device_t dev)
{
struct csa_info *csa;
csa_res *resp;
csa = pcm_getdevinfo(dev);
resp = &csa->res;
csa_active(csa, 1);
/* cs_hardware_init */
csa_stopplaydma(csa);
csa_stopcapturedma(csa);
csa_ac97_resume(csa);
if (csa_startdsp(resp))
return (ENXIO);
/* Enable interrupts on the part. */
if ((csa_readio(resp, BA0_HISR) & HISR_INTENA) == 0)
csa_writeio(resp, BA0_HICR, HICR_IEV | HICR_CHGM);
/* playback interrupt enable */
csa_writemem(resp, BA1_PFIE, csa_readmem(resp, BA1_PFIE) & ~0x0000f03f);
/* capture interrupt enable */
csa_writemem(resp, BA1_CIE,
(csa_readmem(resp, BA1_CIE) & ~0x0000003f) | 0x00000001);
/* cs_restart_part */
csa_setupchan(&csa->pch);
csa_startplaydma(csa);
csa_setupchan(&csa->rch);
csa_startcapturedma(csa);
csa_active(csa, -1);
return 0;
}
static device_method_t pcmcsa_methods[] = {
/* Device interface */
DEVMETHOD(device_probe , pcmcsa_probe ),
DEVMETHOD(device_attach, pcmcsa_attach),
DEVMETHOD(device_detach, pcmcsa_detach),
DEVMETHOD(device_suspend, pcmcsa_suspend),
DEVMETHOD(device_resume, pcmcsa_resume),
{ 0, 0 },
};
static driver_t pcmcsa_driver = {
"pcm",
pcmcsa_methods,
PCM_SOFTC_SIZE,
};
DRIVER_MODULE(snd_csapcm, csa, pcmcsa_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_csapcm, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_DEPEND(snd_csapcm, snd_csa, 1, 1, 1);
MODULE_VERSION(snd_csapcm, 1);