freebsd-dev/sys/dev/sound/pci/emu10kx-pcm.c

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/*-
* Copyright (c) 1999 Cameron Grant <gandalf@vilnya.demon.co.uk>
* Copyright (c) 2003-2007 Yuriy Tsibizov <yuriy.tsibizov@gfk.ru>
* 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, WHETHERIN 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.
*
* $FreeBSD$
*/
#include <sys/param.h>
#include <sys/types.h>
#include <sys/bus.h>
#include <machine/bus.h>
#include <sys/rman.h>
#include <sys/systm.h>
#include <sys/sbuf.h>
#include <sys/queue.h>
#include <sys/systm.h>
#include <sys/lock.h>
#include <sys/mutex.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
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/chip.h>
#include <dev/sound/pcm/sound.h>
#include <dev/sound/pcm/ac97.h>
#include "mixer_if.h"
#include <dev/sound/pci/emu10kx.h>
#include "emu10k1-alsa%diked.h"
struct emu_pcm_pchinfo {
int spd;
int fmt;
unsigned int blksz;
int run;
struct emu_voice *master;
struct emu_voice *slave;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
struct emu_pcm_info *pcm;
int timer;
};
struct emu_pcm_rchinfo {
int spd;
int fmt;
unsigned int blksz;
int run;
uint32_t idxreg;
uint32_t basereg;
uint32_t sizereg;
uint32_t setupreg;
uint32_t irqmask;
uint32_t iprmask;
int ihandle;
struct snd_dbuf *buffer;
struct pcm_channel *channel;
struct emu_pcm_info *pcm;
int timer;
};
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
/* XXX Hardware playback channels */
#define MAX_CHANNELS 4
#if MAX_CHANNELS > 13
#error Too many hardware channels defined. 13 is the maximum
#endif
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
struct emu_pcm_info {
struct mtx *lock;
device_t dev; /* device information */
struct emu_sc_info *card;
struct emu_pcm_pchinfo pch[MAX_CHANNELS]; /* hardware channels */
int pnum; /* next free channel number */
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
struct emu_pcm_rchinfo rch_adc;
struct emu_pcm_rchinfo rch_efx;
struct emu_route rt;
struct emu_route rt_mono;
int route;
int ihandle; /* interrupt handler */
unsigned int bufsz;
int is_emu10k1;
struct ac97_info *codec;
uint32_t ac97_state[0x7F];
kobj_class_t ac97_mixerclass;
uint32_t ac97_recdevs;
uint32_t ac97_playdevs;
struct snd_mixer *sm;
int mch_disabled;
unsigned int emu10k1_volcache[2][2];
};
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
static uint32_t emu_rfmt_adc[] = {
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
};
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
static struct pcmchan_caps emu_reccaps_adc = {
8000, 48000, emu_rfmt_adc, 0
};
static uint32_t emu_rfmt_efx[] = {
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),
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
0
};
static struct pcmchan_caps emu_reccaps_efx_live = {
48000*32, 48000*32, emu_rfmt_efx, 0
};
static struct pcmchan_caps emu_reccaps_efx_audigy = {
48000*64, 48000*64, emu_rfmt_efx, 0
};
static int emu_rates_live[] = {
48000*32
};
static int emu_rates_audigy[] = {
48000*64
};
static uint32_t emu_pfmt[] = {
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_S16_LE, 1, 0),
SND_FORMAT(AFMT_S16_LE, 2, 0),
0
};
static uint32_t emu_pfmt_mono[] = {
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_S16_LE, 1, 0),
0
};
static struct pcmchan_caps emu_playcaps = {4000, 48000, emu_pfmt, 0};
static struct pcmchan_caps emu_playcaps_mono = {4000, 48000, emu_pfmt_mono, 0};
static int emu10k1_adcspeed[8] = {48000, 44100, 32000, 24000, 22050, 16000, 11025, 8000};
/* audigy supports 12kHz. */
static int emu10k2_adcspeed[9] = {48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000};
static uint32_t emu_pcm_intr(void *pcm, uint32_t stat);
static const struct emu_dspmix_props_k1 {
uint8_t present;
uint8_t recdev;
int8_t input;
} dspmix_k1 [SOUND_MIXER_NRDEVICES] = {
/* no mixer device for ac97 */ /* in0 AC97 */
[SOUND_MIXER_DIGITAL1] = {1, 1, 1}, /* in1 CD SPDIF */
/* not connected */ /* in2 (zoom) */
[SOUND_MIXER_DIGITAL2] = {1, 1, 3}, /* in3 toslink */
[SOUND_MIXER_LINE2] = {1, 1, 4}, /* in4 Line-In2 */
[SOUND_MIXER_DIGITAL3] = {1, 1, 5}, /* in5 on-card SPDIF */
[SOUND_MIXER_LINE3] = {1, 1, 6}, /* in6 AUX2 */
/* not connected */ /* in7 */
};
static const struct emu_dspmix_props_k2 {
uint8_t present;
uint8_t recdev;
int8_t input;
} dspmix_k2 [SOUND_MIXER_NRDEVICES] = {
[SOUND_MIXER_VOLUME] = {1, 0, (-1)},
[SOUND_MIXER_PCM] = {1, 0, (-1)},
/* no mixer device */ /* in0 AC97 */
[SOUND_MIXER_DIGITAL1] = {1, 1, 1}, /* in1 CD SPDIF */
[SOUND_MIXER_DIGITAL2] = {1, 1, 2}, /* in2 COAX SPDIF */
/* not connected */ /* in3 */
[SOUND_MIXER_LINE2] = {1, 1, 4}, /* in4 Line-In2 */
[SOUND_MIXER_DIGITAL3] = {1, 1, 5}, /* in5 on-card SPDIF */
[SOUND_MIXER_LINE3] = {1, 1, 6}, /* in6 AUX2 */
/* not connected */ /* in7 */
};
static int
emu_dspmixer_init(struct snd_mixer *m)
{
struct emu_pcm_info *sc;
int i;
int p, r;
p = 0;
r = 0;
sc = mix_getdevinfo(m);
if (sc->route == RT_FRONT) {
/* create submixer for AC97 codec */
if ((sc->ac97_mixerclass != NULL) && (sc->codec != NULL)) {
sc->sm = mixer_create(sc->dev, sc->ac97_mixerclass, sc->codec, "ac97");
if (sc->sm != NULL) {
p = mix_getdevs(sc->sm);
r = mix_getrecdevs(sc->sm);
}
}
sc->ac97_playdevs = p;
sc->ac97_recdevs = r;
}
/* This two are always here */
p |= (1 << SOUND_MIXER_PCM);
p |= (1 << SOUND_MIXER_VOLUME);
if (sc->route == RT_FRONT) {
if (sc->is_emu10k1) {
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (dspmix_k1[i].present)
p |= (1 << i);
if (dspmix_k1[i].recdev)
r |= (1 << i);
}
} else {
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (dspmix_k2[i].present)
p |= (1 << i);
if (dspmix_k2[i].recdev)
r |= (1 << i);
}
}
}
mix_setdevs(m, p);
mix_setrecdevs(m, r);
return (0);
}
static int
emu_dspmixer_uninit(struct snd_mixer *m)
{
struct emu_pcm_info *sc;
int err = 0;
/* drop submixer for AC97 codec */
sc = mix_getdevinfo(m);
if (sc->sm != NULL)
err = mixer_delete(sc->sm);
if (err)
return (err);
sc->sm = NULL;
return (0);
}
static int
emu_dspmixer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
struct emu_pcm_info *sc;
sc = mix_getdevinfo(m);
switch (dev) {
case SOUND_MIXER_VOLUME:
switch (sc->route) {
case RT_FRONT:
if (sc->sm != NULL)
mix_set(sc->sm, dev, left, right);
if (sc->mch_disabled) {
/* In emu10k1 case PCM volume does not affect
sound routed to rear & center/sub (it is connected
to AC97 codec). Calculate it manually. */
/* This really should belong to emu10kx.c */
if (sc->is_emu10k1) {
sc->emu10k1_volcache[0][0] = left;
left = left * sc->emu10k1_volcache[1][0] / 100;
sc->emu10k1_volcache[0][1] = right;
right = right * sc->emu10k1_volcache[1][1] / 100;
}
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
if (!sc->is_emu10k1) {
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
/* XXX side */
}
} /* mch disabled */
break;
case RT_REAR:
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
break;
case RT_CENTER:
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
break;
case RT_SUB:
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
break;
}
break;
case SOUND_MIXER_PCM:
switch (sc->route) {
case RT_FRONT:
if (sc->sm != NULL)
mix_set(sc->sm, dev, left, right);
if (sc->mch_disabled) {
/* See SOUND_MIXER_VOLUME case */
if (sc->is_emu10k1) {
sc->emu10k1_volcache[1][0] = left;
left = left * sc->emu10k1_volcache[0][0] / 100;
sc->emu10k1_volcache[1][1] = right;
right = right * sc->emu10k1_volcache[0][1] / 100;
}
emumix_set_volume(sc->card, M_MASTER_REAR_L, left);
emumix_set_volume(sc->card, M_MASTER_REAR_R, right);
if (!sc->is_emu10k1) {
emumix_set_volume(sc->card, M_MASTER_CENTER, (left+right)/2);
emumix_set_volume(sc->card, M_MASTER_SUBWOOFER, (left+right)/2);
/* XXX side */
}
} /* mch_disabled */
break;
case RT_REAR:
emumix_set_volume(sc->card, M_FX2_REAR_L, left);
emumix_set_volume(sc->card, M_FX3_REAR_R, right);
break;
case RT_CENTER:
emumix_set_volume(sc->card, M_FX4_CENTER, (left+right)/2);
break;
case RT_SUB:
emumix_set_volume(sc->card, M_FX5_SUBWOOFER, (left+right)/2);
break;
}
break;
case SOUND_MIXER_DIGITAL1: /* CD SPDIF, in1 */
emumix_set_volume(sc->card, M_IN1_FRONT_L, left);
emumix_set_volume(sc->card, M_IN1_FRONT_R, right);
break;
case SOUND_MIXER_DIGITAL2:
if (sc->is_emu10k1) {
/* TOSLink, in3 */
emumix_set_volume(sc->card, M_IN3_FRONT_L, left);
emumix_set_volume(sc->card, M_IN3_FRONT_R, right);
} else {
/* COAX SPDIF, in2 */
emumix_set_volume(sc->card, M_IN2_FRONT_L, left);
emumix_set_volume(sc->card, M_IN2_FRONT_R, right);
}
break;
case SOUND_MIXER_LINE2: /* Line-In2, in4 */
emumix_set_volume(sc->card, M_IN4_FRONT_L, left);
emumix_set_volume(sc->card, M_IN4_FRONT_R, right);
break;
case SOUND_MIXER_DIGITAL3: /* on-card SPDIF, in5 */
emumix_set_volume(sc->card, M_IN5_FRONT_L, left);
emumix_set_volume(sc->card, M_IN5_FRONT_R, right);
break;
case SOUND_MIXER_LINE3: /* AUX2, in6 */
emumix_set_volume(sc->card, M_IN6_FRONT_L, left);
emumix_set_volume(sc->card, M_IN6_FRONT_R, right);
break;
default:
if (sc->sm != NULL) {
/* XXX emumix_set_volume is not required here */
emumix_set_volume(sc->card, M_IN0_FRONT_L, 100);
emumix_set_volume(sc->card, M_IN0_FRONT_R, 100);
mix_set(sc->sm, dev, left, right);
} else
device_printf(sc->dev, "mixer error: unknown device %d\n", dev);
}
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
emu_dspmixer_setrecsrc(struct snd_mixer *m, u_int32_t src)
{
struct emu_pcm_info *sc;
int i;
u_int32_t recmask;
int input[8];
sc = mix_getdevinfo(m);
recmask = 0;
for (i=0; i < 8; i++)
input[i]=0;
if (sc->sm != NULL)
if ((src & sc->ac97_recdevs) !=0)
if (mix_setrecsrc(sc->sm, src & sc->ac97_recdevs) == 0) {
recmask |= (src & sc->ac97_recdevs);
/* Recording from AC97 codec.
Enable AC97 route to rec on DSP */
input[0] = 1;
}
if (sc->is_emu10k1) {
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (dspmix_k1[i].recdev)
if ((src & (1 << i)) == ((uint32_t)1 << i)) {
recmask |= (1 << i);
/* enable device i */
input[dspmix_k1[i].input] = 1;
}
}
} else {
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (dspmix_k2[i].recdev)
if ((src & (1 << i)) == ((uint32_t)1 << i)) {
recmask |= (1 << i);
/* enable device i */
input[dspmix_k2[i].input] = 1;
}
}
}
emumix_set_volume(sc->card, M_IN0_REC_L, input[0] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN0_REC_R, input[0] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN1_REC_L, input[1] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN1_REC_R, input[1] == 1 ? 100 : 0);
if (!sc->is_emu10k1) {
emumix_set_volume(sc->card, M_IN2_REC_L, input[2] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN2_REC_R, input[2] == 1 ? 100 : 0);
}
if (sc->is_emu10k1) {
emumix_set_volume(sc->card, M_IN3_REC_L, input[3] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN3_REC_R, input[3] == 1 ? 100 : 0);
}
emumix_set_volume(sc->card, M_IN4_REC_L, input[4] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN4_REC_R, input[4] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN5_REC_L, input[5] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN5_REC_R, input[5] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN6_REC_L, input[6] == 1 ? 100 : 0);
emumix_set_volume(sc->card, M_IN6_REC_R, input[6] == 1 ? 100 : 0);
/* XXX check for K1/k2 differences? */
if ((src & (1 << SOUND_MIXER_PCM)) == (1 << SOUND_MIXER_PCM)) {
emumix_set_volume(sc->card, M_FX0_REC_L, emumix_get_volume(sc->card, M_FX0_FRONT_L));
emumix_set_volume(sc->card, M_FX1_REC_R, emumix_get_volume(sc->card, M_FX1_FRONT_R));
} else {
emumix_set_volume(sc->card, M_FX0_REC_L, 0);
emumix_set_volume(sc->card, M_FX1_REC_R, 0);
}
return (recmask);
}
static kobj_method_t emudspmixer_methods[] = {
KOBJMETHOD(mixer_init, emu_dspmixer_init),
KOBJMETHOD(mixer_uninit, emu_dspmixer_uninit),
KOBJMETHOD(mixer_set, emu_dspmixer_set),
KOBJMETHOD(mixer_setrecsrc, emu_dspmixer_setrecsrc),
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
};
MIXER_DECLARE(emudspmixer);
static int
emu_efxmixer_init(struct snd_mixer *m)
{
mix_setdevs(m, SOUND_MASK_VOLUME);
mix_setrecdevs(m, SOUND_MASK_MONITOR);
return (0);
}
static int
emu_efxmixer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
if (left + right == 200) return (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
static u_int32_t
emu_efxmixer_setrecsrc(struct snd_mixer *m __unused, u_int32_t src __unused)
{
return (SOUND_MASK_MONITOR);
}
static kobj_method_t emuefxmixer_methods[] = {
KOBJMETHOD(mixer_init, emu_efxmixer_init),
KOBJMETHOD(mixer_set, emu_efxmixer_set),
KOBJMETHOD(mixer_setrecsrc, emu_efxmixer_setrecsrc),
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
};
MIXER_DECLARE(emuefxmixer);
/*
* AC97 emulation code for Audigy and later cards.
* Some parts of AC97 codec are not used by hardware, but can be used
* to change some DSP controls via AC97 mixer interface. This includes:
* - master volume controls MASTER_FRONT_[R|L]
* - pcm volume controls FX[0|1]_FRONT_[R|L]
* - rec volume controls MASTER_REC_[R|L]
* We do it because we need to put it under user control....
* We also keep some parts of AC97 disabled to get better sound quality
*/
#define AC97LEFT(x) ((x & 0x7F00)>>8)
#define AC97RIGHT(x) (x & 0x007F)
#define AC97MUTE(x) ((x & 0x8000)>>15)
#define BIT4_TO100(x) (100-(x)*100/(0x0f))
#define BIT6_TO100(x) (100-(x)*100/(0x3f))
#define BIT4_TO255(x) (255-(x)*255/(0x0f))
#define BIT6_TO255(x) (255-(x)*255/(0x3f))
#define V100_TOBIT6(x) (0x3f*(100-x)/100)
#define V100_TOBIT4(x) (0x0f*(100-x)/100)
#define AC97ENCODE(x_muted, x_left, x_right) (((x_muted & 1)<<15) | ((x_left & 0x3f)<<8) | (x_right & 0x3f))
static int
emu_ac97_read_emulation(struct emu_pcm_info *sc, int regno)
{
int use_ac97;
int emulated;
int tmp;
use_ac97 = 1;
emulated = 0;
switch (regno) {
case AC97_MIX_MASTER:
emulated = sc->ac97_state[AC97_MIX_MASTER];
use_ac97 = 0;
break;
case AC97_MIX_PCM:
emulated = sc->ac97_state[AC97_MIX_PCM];
use_ac97 = 0;
break;
case AC97_REG_RECSEL:
emulated = 0x0505;
use_ac97 = 0;
break;
case AC97_MIX_RGAIN:
emulated = sc->ac97_state[AC97_MIX_RGAIN];
use_ac97 = 0;
break;
}
emu_wr(sc->card, AC97ADDRESS, regno, 1);
tmp = emu_rd(sc->card, AC97DATA, 2);
if (use_ac97)
emulated = tmp;
return (emulated);
}
static void
emu_ac97_write_emulation(struct emu_pcm_info *sc, int regno, uint32_t data)
{
int write_ac97;
int left, right;
uint32_t emu_left, emu_right;
int is_mute;
write_ac97 = 1;
left = AC97LEFT(data);
emu_left = BIT6_TO100(left); /* We show us as 6-bit AC97 mixer */
right = AC97RIGHT(data);
emu_right = BIT6_TO100(right);
is_mute = AC97MUTE(data);
if (is_mute)
emu_left = emu_right = 0;
switch (regno) {
/* TODO: reset emulator on AC97_RESET */
case AC97_MIX_MASTER:
emumix_set_volume(sc->card, M_MASTER_FRONT_L, emu_left);
emumix_set_volume(sc->card, M_MASTER_FRONT_R, emu_right);
sc->ac97_state[AC97_MIX_MASTER] = data & (0x8000 | 0x3f3f);
data = 0x8000; /* Mute AC97 main out */
break;
case AC97_MIX_PCM: /* PCM OUT VOL */
emumix_set_volume(sc->card, M_FX0_FRONT_L, emu_left);
emumix_set_volume(sc->card, M_FX1_FRONT_R, emu_right);
sc->ac97_state[AC97_MIX_PCM] = data & (0x8000 | 0x3f3f);
data = 0x8000; /* Mute AC97 PCM out */
break;
case AC97_REG_RECSEL:
/*
* PCM recording source is set to "stereo mix" (labeled "vol"
* in mixer). There is no 'playback' from AC97 codec -
* if you want to hear anything from AC97 you have to _record_
* it. Keep things simple and record "stereo mix".
*/
data = 0x0505;
break;
case AC97_MIX_RGAIN: /* RECORD GAIN */
emu_left = BIT4_TO100(left); /* rgain is 4-bit */
emu_right = BIT4_TO100(right);
emumix_set_volume(sc->card, M_MASTER_REC_L, 100-emu_left);
emumix_set_volume(sc->card, M_MASTER_REC_R, 100-emu_right);
/*
* Record gain on AC97 should stay zero to get AC97 sound on
* AC97_[RL] connectors on EMU10K2 chip. AC97 on Audigy is not
* directly connected to any output, only to EMU10K2 chip Use
* this control to set AC97 mix volume inside EMU10K2 chip
*/
sc->ac97_state[AC97_MIX_RGAIN] = data & (0x8000 | 0x0f0f);
data = 0x0000;
break;
}
if (write_ac97) {
emu_wr(sc->card, AC97ADDRESS, regno, 1);
emu_wr(sc->card, AC97DATA, data, 2);
}
}
static int
emu_erdcd(kobj_t obj __unused, void *devinfo, int regno)
{
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
return (emu_ac97_read_emulation(sc, regno));
}
static int
emu_ewrcd(kobj_t obj __unused, void *devinfo, int regno, uint32_t data)
{
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
emu_ac97_write_emulation(sc, regno, data);
return (0);
}
static kobj_method_t emu_eac97_methods[] = {
KOBJMETHOD(ac97_read, emu_erdcd),
KOBJMETHOD(ac97_write, emu_ewrcd),
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(emu_eac97);
/* real ac97 codec */
static int
emu_rdcd(kobj_t obj __unused, void *devinfo, int regno)
{
int rd;
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
KASSERT(sc->card != NULL, ("emu_rdcd: no soundcard"));
emu_wr(sc->card, AC97ADDRESS, regno, 1);
rd = emu_rd(sc->card, AC97DATA, 2);
return (rd);
}
static int
emu_wrcd(kobj_t obj __unused, void *devinfo, int regno, uint32_t data)
{
struct emu_pcm_info *sc = (struct emu_pcm_info *)devinfo;
KASSERT(sc->card != NULL, ("emu_wrcd: no soundcard"));
emu_wr(sc->card, AC97ADDRESS, regno, 1);
emu_wr(sc->card, AC97DATA, data, 2);
return (0);
}
static kobj_method_t emu_ac97_methods[] = {
KOBJMETHOD(ac97_read, emu_rdcd),
KOBJMETHOD(ac97_write, emu_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(emu_ac97);
static int
emu_k1_recval(int speed)
{
int val;
val = 0;
while ((val < 7) && (speed < emu10k1_adcspeed[val]))
val++;
return (val);
}
static int
emu_k2_recval(int speed)
{
int val;
val = 0;
while ((val < 8) && (speed < emu10k2_adcspeed[val]))
val++;
return (val);
}
static void *
emupchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
{
struct emu_pcm_info *sc = devinfo;
struct emu_pcm_pchinfo *ch;
void *r;
KASSERT(dir == PCMDIR_PLAY, ("emupchan_init: bad direction"));
KASSERT(sc->card != NULL, ("empchan_init: no soundcard"));
if (sc->pnum >= MAX_CHANNELS)
return (NULL);
ch = &(sc->pch[sc->pnum++]);
ch->buffer = b;
ch->pcm = sc;
ch->channel = c;
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch->blksz = sc->bufsz;
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
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
ch->spd = 8000;
ch->master = emu_valloc(sc->card);
/*
* XXX we have to allocate slave even for mono channel until we
* fix emu_vfree to handle this case.
*/
ch->slave = emu_valloc(sc->card);
ch->timer = emu_timer_create(sc->card);
r = (emu_vinit(sc->card, ch->master, ch->slave, EMU_PLAY_BUFSZ, ch->buffer)) ? NULL : ch;
return (r);
}
static int
emupchan_free(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
emu_timer_clear(sc->card, ch->timer);
if (ch->slave != NULL)
emu_vfree(sc->card, ch->slave);
emu_vfree(sc->card, ch->master);
return (0);
}
static int
emupchan_setformat(kobj_t obj __unused, void *c_devinfo, uint32_t format)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
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 uint32_t
emupchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
ch->spd = speed;
return (ch->spd);
}
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 uint32_t
emupchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
if (blocksize > ch->pcm->bufsz)
blocksize = ch->pcm->bufsz;
snd_mtxlock(sc->lock);
ch->blksz = blocksize;
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
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
snd_mtxunlock(sc->lock);
return (ch->blksz);
}
static int
emupchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
if (!PCMTRIG_COMMON(go))
return (0);
snd_mtxlock(sc->lock); /* XXX can we trigger on parallel threads ? */
if (go == PCMTRIG_START) {
emu_vsetup(ch->master, ch->fmt, ch->spd);
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)
emu_vroute(sc->card, &(sc->rt), ch->master);
else
emu_vroute(sc->card, &(sc->rt_mono), ch->master);
emu_vwrite(sc->card, ch->master);
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
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
emu_timer_enable(sc->card, ch->timer, 1);
}
/* PCM interrupt handler will handle PCMTRIG_STOP event */
ch->run = (go == PCMTRIG_START) ? 1 : 0;
emu_vtrigger(sc->card, ch->master, ch->run);
snd_mtxunlock(sc->lock);
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 uint32_t
emupchan_getptr(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
int r;
r = emu_vpos(sc->card, ch->master);
return (r);
}
static struct pcmchan_caps *
emupchan_getcaps(kobj_t obj __unused, void *c_devinfo __unused)
{
struct emu_pcm_pchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
switch (sc->route) {
case RT_FRONT:
/* FALLTHROUGH */
case RT_REAR:
/* FALLTHROUGH */
case RT_SIDE:
return (&emu_playcaps);
break;
case RT_CENTER:
/* FALLTHROUGH */
case RT_SUB:
return (&emu_playcaps_mono);
break;
}
return (NULL);
}
static kobj_method_t emupchan_methods[] = {
KOBJMETHOD(channel_init, emupchan_init),
KOBJMETHOD(channel_free, emupchan_free),
KOBJMETHOD(channel_setformat, emupchan_setformat),
KOBJMETHOD(channel_setspeed, emupchan_setspeed),
KOBJMETHOD(channel_setblocksize, emupchan_setblocksize),
KOBJMETHOD(channel_trigger, emupchan_trigger),
KOBJMETHOD(channel_getptr, emupchan_getptr),
KOBJMETHOD(channel_getcaps, emupchan_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(emupchan);
static void *
emurchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
{
struct emu_pcm_info *sc = devinfo;
struct emu_pcm_rchinfo *ch;
KASSERT(dir == PCMDIR_REC, ("emurchan_init: bad direction"));
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch = &sc->rch_adc;
ch->buffer = b;
ch->pcm = sc;
ch->channel = c;
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch->blksz = sc->bufsz / 2; /* We rise interrupt for half-full buffer */
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
ch->fmt = SND_FORMAT(AFMT_U8, 1, 0);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch->spd = 8000;
ch->idxreg = sc->is_emu10k1 ? ADCIDX : A_ADCIDX;
ch->basereg = ADCBA;
ch->sizereg = ADCBS;
ch->setupreg = ADCCR;
ch->irqmask = INTE_ADCBUFENABLE;
ch->iprmask = IPR_ADCBUFFULL | IPR_ADCBUFHALFFULL;
if (sndbuf_alloc(ch->buffer, emu_gettag(sc->card), 0, sc->bufsz) != 0)
return (NULL);
else {
ch->timer = emu_timer_create(sc->card);
emu_wrptr(sc->card, 0, ch->basereg, sndbuf_getbufaddr(ch->buffer));
emu_wrptr(sc->card, 0, ch->sizereg, 0); /* off */
return (ch);
}
}
static int
emurchan_free(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
emu_timer_clear(sc->card, ch->timer);
return (0);
}
static int
emurchan_setformat(kobj_t obj __unused, void *c_devinfo, uint32_t format)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
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 uint32_t
emurchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
if (ch->pcm->is_emu10k1) {
speed = emu10k1_adcspeed[emu_k1_recval(speed)];
} else {
speed = emu10k2_adcspeed[emu_k2_recval(speed)];
}
ch->spd = speed;
return (ch->spd);
}
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 uint32_t
emurchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
ch->blksz = blocksize;
/*
* If blocksize is less than half of buffer size we will not get
* BUFHALFFULL interrupt in time and channel will need to generate
* (and use) timer interrupts. Otherwise channel will be marked dead.
*/
if (ch->blksz < (ch->pcm->bufsz / 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
emu_timer_set(sc->card, ch->timer, ch->blksz / sndbuf_getalign(ch->buffer));
emu_timer_enable(sc->card, ch->timer, 1);
} else {
emu_timer_enable(sc->card, ch->timer, 0);
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
return (ch->blksz);
}
static int
emurchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
uint32_t val, sz;
if (!PCMTRIG_COMMON(go))
return (0);
switch (sc->bufsz) {
case 4096:
sz = ADCBS_BUFSIZE_4096;
break;
case 8192:
sz = ADCBS_BUFSIZE_8192;
break;
case 16384:
sz = ADCBS_BUFSIZE_16384;
break;
case 32768:
sz = ADCBS_BUFSIZE_32768;
break;
case 65536:
sz = ADCBS_BUFSIZE_65536;
break;
default:
sz = ADCBS_BUFSIZE_4096;
}
snd_mtxlock(sc->lock);
switch (go) {
case PCMTRIG_START:
ch->run = 1;
emu_wrptr(sc->card, 0, ch->sizereg, sz);
val = sc->is_emu10k1 ? ADCCR_LCHANENABLE : A_ADCCR_LCHANENABLE;
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)
val |= sc->is_emu10k1 ? ADCCR_RCHANENABLE : A_ADCCR_RCHANENABLE;
val |= sc->is_emu10k1 ? emu_k1_recval(ch->spd) : emu_k2_recval(ch->spd);
emu_wrptr(sc->card, 0, ch->setupreg, 0);
emu_wrptr(sc->card, 0, ch->setupreg, val);
ch->ihandle = emu_intr_register(sc->card, ch->irqmask, ch->iprmask, &emu_pcm_intr, sc);
break;
case PCMTRIG_STOP:
/* FALLTHROUGH */
case PCMTRIG_ABORT:
ch->run = 0;
emu_wrptr(sc->card, 0, ch->sizereg, 0);
if (ch->setupreg)
emu_wrptr(sc->card, 0, ch->setupreg, 0);
(void)emu_intr_unregister(sc->card, ch->ihandle);
break;
case PCMTRIG_EMLDMAWR:
/* FALLTHROUGH */
case PCMTRIG_EMLDMARD:
/* FALLTHROUGH */
default:
break;
}
snd_mtxunlock(sc->lock);
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 uint32_t
emurchan_getptr(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
int r;
r = emu_rdptr(sc->card, 0, ch->idxreg) & 0x0000ffff;
return (r);
}
static struct pcmchan_caps *
emurchan_getcaps(kobj_t obj __unused, void *c_devinfo __unused)
{
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
return (&emu_reccaps_adc);
}
static kobj_method_t emurchan_methods[] = {
KOBJMETHOD(channel_init, emurchan_init),
KOBJMETHOD(channel_free, emurchan_free),
KOBJMETHOD(channel_setformat, emurchan_setformat),
KOBJMETHOD(channel_setspeed, emurchan_setspeed),
KOBJMETHOD(channel_setblocksize, emurchan_setblocksize),
KOBJMETHOD(channel_trigger, emurchan_trigger),
KOBJMETHOD(channel_getptr, emurchan_getptr),
KOBJMETHOD(channel_getcaps, emurchan_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(emurchan);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
static void *
emufxrchan_init(kobj_t obj __unused, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir __unused)
{
struct emu_pcm_info *sc = devinfo;
struct emu_pcm_rchinfo *ch;
KASSERT(dir == PCMDIR_REC, ("emurchan_init: bad direction"));
if (sc == NULL) return (NULL);
ch = &(sc->rch_efx);
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
ch->fmt = SND_FORMAT(AFMT_S16_LE, 1, 0);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch->spd = sc->is_emu10k1 ? 48000*32 : 48000 * 64;
ch->idxreg = FXIDX;
ch->basereg = FXBA;
ch->sizereg = FXBS;
ch->irqmask = INTE_EFXBUFENABLE;
ch->iprmask = IPR_EFXBUFFULL | IPR_EFXBUFHALFFULL;
ch->buffer = b;
ch->pcm = sc;
ch->channel = c;
ch->blksz = sc->bufsz / 2;
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
if (sndbuf_alloc(ch->buffer, emu_gettag(sc->card), 0, sc->bufsz) != 0)
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
return (NULL);
else {
emu_wrptr(sc->card, 0, ch->basereg, sndbuf_getbufaddr(ch->buffer));
emu_wrptr(sc->card, 0, ch->sizereg, 0); /* off */
return (ch);
}
}
static int
emufxrchan_setformat(kobj_t obj __unused, void *c_devinfo __unused, uint32_t format)
{
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 (format == SND_FORMAT(AFMT_S16_LE, 1, 0)) return (0);
return (EINVAL);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +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
static uint32_t
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
emufxrchan_setspeed(kobj_t obj __unused, void *c_devinfo, uint32_t speed)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
/* FIXED RATE CHANNEL */
return (ch->spd);
}
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 uint32_t
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
emufxrchan_setblocksize(kobj_t obj __unused, void *c_devinfo, uint32_t blocksize)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
ch->blksz = blocksize;
/*
* XXX If blocksize is less than half of buffer size we will not get
* interrupt in time and channel will die due to interrupt timeout.
* This should not happen with FX rchan, because it will fill buffer
* very fast (64K buffer is 0.021seconds on Audigy).
*/
if (ch->blksz < (ch->pcm->bufsz / 2))
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
ch->blksz = ch->pcm->bufsz / 2;
return (ch->blksz);
}
static int
emufxrchan_trigger(kobj_t obj __unused, void *c_devinfo, int go)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
uint32_t sz;
if (!PCMTRIG_COMMON(go))
return (0);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
switch (sc->bufsz) {
case 4096:
sz = ADCBS_BUFSIZE_4096;
break;
case 8192:
sz = ADCBS_BUFSIZE_8192;
break;
case 16384:
sz = ADCBS_BUFSIZE_16384;
break;
case 32768:
sz = ADCBS_BUFSIZE_32768;
break;
case 65536:
sz = ADCBS_BUFSIZE_65536;
break;
default:
sz = ADCBS_BUFSIZE_4096;
}
snd_mtxlock(sc->lock);
switch (go) {
case PCMTRIG_START:
ch->run = 1;
emu_wrptr(sc->card, 0, ch->sizereg, sz);
ch->ihandle = emu_intr_register(sc->card, ch->irqmask, ch->iprmask, &emu_pcm_intr, sc);
/*
* SB Live! is limited to 32 mono channels. Audigy
* has 64 mono channels. Channels are enabled
* by setting a bit in A_FXWC[1|2] registers.
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
*/
/* XXX there is no way to demultiplex this streams for now */
if (sc->is_emu10k1) {
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
emu_wrptr(sc->card, 0, FXWC, 0xffffffff);
} else {
emu_wrptr(sc->card, 0, A_FXWC1, 0xffffffff);
emu_wrptr(sc->card, 0, A_FXWC2, 0xffffffff);
}
break;
case PCMTRIG_STOP:
/* FALLTHROUGH */
case PCMTRIG_ABORT:
ch->run = 0;
if (sc->is_emu10k1) {
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
emu_wrptr(sc->card, 0, FXWC, 0x0);
} else {
emu_wrptr(sc->card, 0, A_FXWC1, 0x0);
emu_wrptr(sc->card, 0, A_FXWC2, 0x0);
}
emu_wrptr(sc->card, 0, ch->sizereg, 0);
(void)emu_intr_unregister(sc->card, ch->ihandle);
break;
case PCMTRIG_EMLDMAWR:
/* FALLTHROUGH */
case PCMTRIG_EMLDMARD:
/* FALLTHROUGH */
default:
break;
}
snd_mtxunlock(sc->lock);
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 uint32_t
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
emufxrchan_getptr(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
int r;
r = emu_rdptr(sc->card, 0, ch->idxreg) & 0x0000ffff;
return (r);
}
static struct pcmchan_caps *
emufxrchan_getcaps(kobj_t obj __unused, void *c_devinfo)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
if (sc->is_emu10k1)
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
return (&emu_reccaps_efx_live);
return (&emu_reccaps_efx_audigy);
}
static int
emufxrchan_getrates(kobj_t obj __unused, void *c_devinfo, int **rates)
{
struct emu_pcm_rchinfo *ch = c_devinfo;
struct emu_pcm_info *sc = ch->pcm;
if (sc->is_emu10k1)
*rates = emu_rates_live;
else
*rates = emu_rates_audigy;
return 1;
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
static kobj_method_t emufxrchan_methods[] = {
KOBJMETHOD(channel_init, emufxrchan_init),
KOBJMETHOD(channel_setformat, emufxrchan_setformat),
KOBJMETHOD(channel_setspeed, emufxrchan_setspeed),
KOBJMETHOD(channel_setblocksize, emufxrchan_setblocksize),
KOBJMETHOD(channel_trigger, emufxrchan_trigger),
KOBJMETHOD(channel_getptr, emufxrchan_getptr),
KOBJMETHOD(channel_getcaps, emufxrchan_getcaps),
KOBJMETHOD(channel_getrates, emufxrchan_getrates),
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
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
};
CHANNEL_DECLARE(emufxrchan);
static uint32_t
emu_pcm_intr(void *pcm, uint32_t stat)
{
struct emu_pcm_info *sc = (struct emu_pcm_info *)pcm;
uint32_t ack;
int i;
ack = 0;
snd_mtxlock(sc->lock);
if (stat & IPR_INTERVALTIMER) {
ack |= IPR_INTERVALTIMER;
for (i = 0; i < MAX_CHANNELS; i++)
if (sc->pch[i].channel) {
if (sc->pch[i].run == 1) {
snd_mtxunlock(sc->lock);
chn_intr(sc->pch[i].channel);
snd_mtxlock(sc->lock);
} else
emu_timer_enable(sc->card, sc->pch[i].timer, 0);
}
/* ADC may install timer to get low-latency interrupts */
if ((sc->rch_adc.channel) && (sc->rch_adc.run)) {
snd_mtxunlock(sc->lock);
chn_intr(sc->rch_adc.channel);
snd_mtxlock(sc->lock);
}
/*
* EFX does not use timer, because it will fill
* buffer at least 32x times faster than ADC.
*/
}
if (stat & (IPR_ADCBUFFULL | IPR_ADCBUFHALFFULL)) {
ack |= stat & (IPR_ADCBUFFULL | IPR_ADCBUFHALFFULL);
if (sc->rch_adc.channel) {
snd_mtxunlock(sc->lock);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
chn_intr(sc->rch_adc.channel);
snd_mtxlock(sc->lock);
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
}
if (stat & (IPR_EFXBUFFULL | IPR_EFXBUFHALFFULL)) {
ack |= stat & (IPR_EFXBUFFULL | IPR_EFXBUFHALFFULL);
if (sc->rch_efx.channel) {
snd_mtxunlock(sc->lock);
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
chn_intr(sc->rch_efx.channel);
snd_mtxlock(sc->lock);
}
}
snd_mtxunlock(sc->lock);
return (ack);
}
static int
emu_pcm_init(struct emu_pcm_info *sc)
{
sc->bufsz = pcm_getbuffersize(sc->dev, EMUPAGESIZE, EMU_REC_BUFSZ, EMU_MAX_BUFSZ);
return (0);
}
static int
emu_pcm_uninit(struct emu_pcm_info *sc __unused)
{
return (0);
}
static int
emu_pcm_probe(device_t dev)
{
uintptr_t func, route, r;
const char *rt;
char buffer[255];
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_FUNC, &func);
if (func != SCF_PCM)
return (ENXIO);
rt = "UNKNOWN";
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ROUTE, &route);
switch (route) {
case RT_FRONT:
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
rt = "front";
break;
case RT_REAR:
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
rt = "rear";
break;
case RT_CENTER:
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
rt = "center";
break;
case RT_SUB:
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
rt = "subwoofer";
break;
case RT_SIDE:
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
rt = "side";
break;
case RT_MCHRECORD:
rt = "multichannel recording";
break;
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
snprintf(buffer, 255, "EMU10Kx DSP %s PCM interface", rt);
device_set_desc_copy(dev, buffer);
return (0);
}
static int
emu_pcm_attach(device_t dev)
{
struct emu_pcm_info *sc;
unsigned int i;
char status[SND_STATUSLEN];
uint32_t inte, ipr;
uintptr_t route, r, ivar;
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
sc->card = (struct emu_sc_info *)(device_get_softc(device_get_parent(dev)));
if (sc->card == NULL) {
device_printf(dev, "cannot get bridge conf\n");
free(sc, M_DEVBUF);
return (ENXIO);
}
sc->lock = snd_mtxcreate(device_get_nameunit(dev), "snd_emu10kx pcm softc");
sc->dev = dev;
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ISEMU10K1, &ivar);
sc->is_emu10k1 = ivar ? 1 : 0;
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_MCH_DISABLED, &ivar);
sc->mch_disabled = ivar ? 1 : 0;
sc->codec = NULL;
for (i = 0; i < 8; i++) {
sc->rt.routing_left[i] = i;
sc->rt.amounts_left[i] = 0x00;
sc->rt.routing_right[i] = i;
sc->rt.amounts_right[i] = 0x00;
}
for (i = 0; i < 8; i++) {
sc->rt_mono.routing_left[i] = i;
sc->rt_mono.amounts_left[i] = 0x00;
sc->rt_mono.routing_right[i] = i;
sc->rt_mono.amounts_right[i] = 0x00;
}
sc->emu10k1_volcache[0][0] = 75;
sc->emu10k1_volcache[1][0] = 75;
sc->emu10k1_volcache[0][1] = 75;
sc->emu10k1_volcache[1][1] = 75;
r = BUS_READ_IVAR(device_get_parent(dev), dev, EMU_VAR_ROUTE, &route);
sc->route = route;
switch (route) {
case RT_FRONT:
sc->rt.amounts_left[0] = 0xff;
sc->rt.amounts_right[1] = 0xff;
sc->rt_mono.amounts_left[0] = 0xff;
sc->rt_mono.amounts_left[1] = 0xff;
if (sc->is_emu10k1)
sc->codec = AC97_CREATE(dev, sc, emu_ac97);
else
sc->codec = AC97_CREATE(dev, sc, emu_eac97);
sc->ac97_mixerclass = NULL;
if (sc->codec != NULL)
sc->ac97_mixerclass = ac97_getmixerclass();
if (mixer_init(dev, &emudspmixer_class, sc)) {
device_printf(dev, "failed to initialize DSP mixer\n");
goto bad;
}
break;
case RT_REAR:
sc->rt.amounts_left[2] = 0xff;
sc->rt.amounts_right[3] = 0xff;
sc->rt_mono.amounts_left[2] = 0xff;
sc->rt_mono.amounts_left[3] = 0xff;
if (mixer_init(dev, &emudspmixer_class, sc)) {
device_printf(dev, "failed to initialize mixer\n");
goto bad;
}
break;
case RT_CENTER:
sc->rt.amounts_left[4] = 0xff;
sc->rt_mono.amounts_left[4] = 0xff;
if (mixer_init(dev, &emudspmixer_class, sc)) {
device_printf(dev, "failed to initialize mixer\n");
goto bad;
}
break;
case RT_SUB:
sc->rt.amounts_left[5] = 0xff;
sc->rt_mono.amounts_left[5] = 0xff;
if (mixer_init(dev, &emudspmixer_class, sc)) {
device_printf(dev, "failed to initialize mixer\n");
goto bad;
}
break;
case RT_SIDE:
sc->rt.amounts_left[6] = 0xff;
sc->rt.amounts_right[7] = 0xff;
sc->rt_mono.amounts_left[6] = 0xff;
sc->rt_mono.amounts_left[7] = 0xff;
if (mixer_init(dev, &emudspmixer_class, sc)) {
device_printf(dev, "failed to initialize mixer\n");
goto bad;
}
break;
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
case RT_MCHRECORD:
if (mixer_init(dev, &emuefxmixer_class, sc)) {
device_printf(dev, "failed to initialize EFX mixer\n");
goto bad;
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
break;
default:
device_printf(dev, "invalid default route\n");
goto bad;
}
inte = INTE_INTERVALTIMERENB;
ipr = IPR_INTERVALTIMER; /* Used by playback & ADC */
sc->ihandle = emu_intr_register(sc->card, inte, ipr, &emu_pcm_intr, sc);
if (emu_pcm_init(sc) == -1) {
device_printf(dev, "unable to initialize PCM part of the card\n");
goto bad;
}
/*
* We don't register interrupt handler with snd_setup_intr
* in pcm device. Mark pcm device as MPSAFE manually.
*/
pcm_setflags(dev, pcm_getflags(dev) | SD_F_MPSAFE);
/* XXX we should better get number of available channels from parent */
if (pcm_register(dev, sc, (route == RT_FRONT) ? MAX_CHANNELS : 1, (route == RT_FRONT) ? 1 : 0)) {
device_printf(dev, "can't register PCM channels!\n");
goto bad;
}
sc->pnum = 0;
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
if (route != RT_MCHRECORD)
pcm_addchan(dev, PCMDIR_PLAY, &emupchan_class, sc);
if (route == RT_FRONT) {
for (i = 1; i < MAX_CHANNELS; i++)
pcm_addchan(dev, PCMDIR_PLAY, &emupchan_class, sc);
pcm_addchan(dev, PCMDIR_REC, &emurchan_class, sc);
}
Extend the emu10kx driver. With the words of the author: ---snip--- New features: 1. Optional multichannel recording (32 channels on Live!, 64 channels on Audigy). All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.. Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream. Channel map: SB Live! (4.0/5.1) offset (words) substream 0x00 Front L 0x01 Front R 0x02 Digital Front L 0x03 Digital Front R 0x04 Digital Center 0x05 Digital Sub 0x06 Headphones L 0x07 Headphones R 0x08 Rear L 0x09 Rear R 0x0A ADC (multi-rate recording) L 0x0B ADC (multi-rate recording) R 0x0C unused 0x0D unused 0x0E unused 0x0F unused 0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0) 0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0) 0x12..-0x1F dummy Audigy / Audigy 2 / Audigy 2 Value / Audigy 4 offset (words) substream 0x00 Digital Front L 0x01 Digital Front R 0x02 Digital Center 0x03 Digital Sub 0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards) 0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards) 0x06 Digital Rear L 0x07 Digital Rear R 0x08 Front L 0x09 Front R 0x0A Center 0x0B Sub 0x0C Side L 0x0D Side R 0x0E Rear L 0x0F Rear R 0x10 output to AC97 input L (muted) 0x11 output to AC97 input R (muted) 0x12 unused 0x13 unused 0x14 unused 0x15 unused 0x16 ADC (multi-rate recording) L 0x17 ADC (multi-rate recording) R 0x18 unused 0x19 unused 0x1A unused 0x1B unused 0x1C unused 0x1D unused 0x1E unused 0x1F unused 0x20..0x3F dummy Fixes: 1. Do not assign negative values to variables used to index emu_cards array. This array was never accessed when index is negative, but Alexander (netchild@) told me that Coverity does not like it. After this change emu_cards[0] should never be used to identify valid sound card. 2. Fix off-by-one errors in interrupt manager. Add more checks there. 3. Fixes to sound buffering code now allows driver to use large playback buffers. 4. Fix memory allocation bug when multichannel recording is not enabled. 5. Fix interrupt timeout when recording with low bitrate (8kHz). Hardware: 1. Add one more known Audigy ZS card to list. Add two cards with PCI IDs betwen old known cards and new one. Other changes: 1. Do not use ALL CAPS in messages. Incomplete code: 1. Automute S/PDIF when S/PDIF signal is lost. Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only). ---snip--- This commits enables a little bit of debugging output when the driver is loaded as a module. I did a cross-build test for amd64. The code has some style issues, this will be addressed later. The multichannel recording part is some work in progress to allow playing around with it until the generic sound code is better able to handle multichannel streams. This is supposed to fix CID: 171187 Found by: Coverity Prevent Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
2007-01-06 18:59:35 +00:00
if (route == RT_MCHRECORD)
pcm_addchan(dev, PCMDIR_REC, &emufxrchan_class, sc);
snprintf(status, SND_STATUSLEN, "on %s", device_get_nameunit(device_get_parent(dev)));
pcm_setstatus(dev, status);
return (0);
bad:
if (sc->codec)
ac97_destroy(sc->codec);
if (sc->lock)
snd_mtxfree(sc->lock);
free(sc, M_DEVBUF);
return (ENXIO);
}
static int
emu_pcm_detach(device_t dev)
{
int r;
struct emu_pcm_info *sc;
sc = pcm_getdevinfo(dev);
r = pcm_unregister(dev);
if (r) return (r);
emu_pcm_uninit(sc);
if (sc->lock)
snd_mtxfree(sc->lock);
free(sc, M_DEVBUF);
return (0);
}
static device_method_t emu_pcm_methods[] = {
DEVMETHOD(device_probe, emu_pcm_probe),
DEVMETHOD(device_attach, emu_pcm_attach),
DEVMETHOD(device_detach, emu_pcm_detach),
{0, 0}
};
static driver_t emu_pcm_driver = {
"pcm",
emu_pcm_methods,
PCM_SOFTC_SIZE,
NULL,
0,
NULL
};
DRIVER_MODULE(snd_emu10kx_pcm, emu10kx, emu_pcm_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_emu10kx_pcm, snd_emu10kx, SND_EMU10KX_MINVER, SND_EMU10KX_PREFVER, SND_EMU10KX_MAXVER);
MODULE_DEPEND(snd_emu10kx_pcm, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
MODULE_VERSION(snd_emu10kx_pcm, SND_EMU10KX_PREFVER);