90da2b2859
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.
243 lines
5.9 KiB
C
243 lines
5.9 KiB
C
/* $FreeBSD$ */
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/*-
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* Copyright (c) 2000-2002 Hiroyuki Aizu <aizu@navi.org>
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* Copyright (c) 2006 Hans Petter Selasky
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*/
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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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#include <dev/sound/pcm/sound.h>
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#include <dev/sound/chip.h>
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#include <dev/sound/usb/uaudio.h>
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#include "mixer_if.h"
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/************************************************************/
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static void *
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ua_chan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b, struct pcm_channel *c, int dir)
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{
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return (uaudio_chan_init(devinfo, b, c, dir));
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}
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static int
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ua_chan_free(kobj_t obj, void *data)
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{
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return (uaudio_chan_free(data));
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}
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static int
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ua_chan_setformat(kobj_t obj, void *data, uint32_t format)
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{
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/*
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* At this point, no need to query as we
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* shouldn't select an unsorted format
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*/
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return (uaudio_chan_set_param_format(data, format));
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}
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static uint32_t
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ua_chan_setspeed(kobj_t obj, void *data, uint32_t speed)
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{
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return (uaudio_chan_set_param_speed(data, speed));
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}
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static uint32_t
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ua_chan_setblocksize(kobj_t obj, void *data, uint32_t blocksize)
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{
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return (uaudio_chan_set_param_blocksize(data, blocksize));
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}
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static int
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ua_chan_setfragments(kobj_t obj, void *data, uint32_t blocksize, uint32_t blockcount)
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{
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return (uaudio_chan_set_param_fragments(data, blocksize, blockcount));
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}
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static int
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ua_chan_trigger(kobj_t obj, void *data, int go)
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{
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if (!PCMTRIG_COMMON(go)) {
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return (0);
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}
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if (go == PCMTRIG_START) {
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return (uaudio_chan_start(data));
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} else {
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return (uaudio_chan_stop(data));
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}
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}
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static uint32_t
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ua_chan_getptr(kobj_t obj, void *data)
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{
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return (uaudio_chan_getptr(data));
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}
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static struct pcmchan_caps *
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ua_chan_getcaps(kobj_t obj, void *data)
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{
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return (uaudio_chan_getcaps(data));
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}
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static struct pcmchan_matrix *
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ua_chan_getmatrix(kobj_t obj, void *data, uint32_t format)
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{
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return (uaudio_chan_getmatrix(data, format));
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}
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static kobj_method_t ua_chan_methods[] = {
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KOBJMETHOD(channel_init, ua_chan_init),
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KOBJMETHOD(channel_free, ua_chan_free),
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KOBJMETHOD(channel_setformat, ua_chan_setformat),
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KOBJMETHOD(channel_setspeed, ua_chan_setspeed),
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KOBJMETHOD(channel_setblocksize, ua_chan_setblocksize),
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KOBJMETHOD(channel_setfragments, ua_chan_setfragments),
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KOBJMETHOD(channel_trigger, ua_chan_trigger),
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KOBJMETHOD(channel_getptr, ua_chan_getptr),
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KOBJMETHOD(channel_getcaps, ua_chan_getcaps),
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KOBJMETHOD(channel_getmatrix, ua_chan_getmatrix),
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KOBJMETHOD_END
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};
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CHANNEL_DECLARE(ua_chan);
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/************************************************************/
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static int
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ua_mixer_init(struct snd_mixer *m)
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{
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return (uaudio_mixer_init_sub(mix_getdevinfo(m), m));
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}
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static int
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ua_mixer_set(struct snd_mixer *m, unsigned type, unsigned left, unsigned right)
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{
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struct mtx *mtx = mixer_get_lock(m);
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uint8_t do_unlock;
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if (mtx_owned(mtx)) {
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do_unlock = 0;
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} else {
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do_unlock = 1;
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mtx_lock(mtx);
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}
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uaudio_mixer_set(mix_getdevinfo(m), type, left, right);
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if (do_unlock) {
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mtx_unlock(mtx);
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}
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return (left | (right << 8));
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}
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static uint32_t
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ua_mixer_setrecsrc(struct snd_mixer *m, uint32_t src)
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{
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struct mtx *mtx = mixer_get_lock(m);
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int retval;
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uint8_t do_unlock;
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if (mtx_owned(mtx)) {
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do_unlock = 0;
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} else {
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do_unlock = 1;
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mtx_lock(mtx);
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}
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retval = uaudio_mixer_setrecsrc(mix_getdevinfo(m), src);
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if (do_unlock) {
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mtx_unlock(mtx);
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}
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return (retval);
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}
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static int
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ua_mixer_uninit(struct snd_mixer *m)
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{
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return (uaudio_mixer_uninit_sub(mix_getdevinfo(m)));
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}
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static kobj_method_t ua_mixer_methods[] = {
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KOBJMETHOD(mixer_init, ua_mixer_init),
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KOBJMETHOD(mixer_uninit, ua_mixer_uninit),
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KOBJMETHOD(mixer_set, ua_mixer_set),
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KOBJMETHOD(mixer_setrecsrc, ua_mixer_setrecsrc),
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KOBJMETHOD_END
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};
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MIXER_DECLARE(ua_mixer);
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/************************************************************/
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static int
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ua_probe(device_t dev)
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{
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struct sndcard_func *func;
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/* the parent device has already been probed */
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func = device_get_ivars(dev);
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if ((func == NULL) ||
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(func->func != SCF_PCM)) {
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return (ENXIO);
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}
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device_set_desc(dev, "USB audio");
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return (BUS_PROBE_DEFAULT);
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}
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static int
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ua_attach(device_t dev)
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{
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return (uaudio_attach_sub(dev, &ua_mixer_class, &ua_chan_class));
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}
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static int
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ua_detach(device_t dev)
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{
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return (uaudio_detach_sub(dev));
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}
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/************************************************************/
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static device_method_t ua_pcm_methods[] = {
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/* Device interface */
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DEVMETHOD(device_probe, ua_probe),
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DEVMETHOD(device_attach, ua_attach),
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DEVMETHOD(device_detach, ua_detach),
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{0, 0}
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};
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static driver_t ua_pcm_driver = {
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"pcm",
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ua_pcm_methods,
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PCM_SOFTC_SIZE,
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};
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DRIVER_MODULE(ua_pcm, uaudio, ua_pcm_driver, pcm_devclass, 0, 0);
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MODULE_DEPEND(ua_pcm, uaudio, 1, 1, 1);
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MODULE_DEPEND(ua_pcm, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
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MODULE_VERSION(ua_pcm, 1);
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