e3faadaafe
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.
118 lines
3.4 KiB
C
118 lines
3.4 KiB
C
/*-
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* Copyright (c) 2002 Orion Hodson
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* All rights reserved.
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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/pcm/ac97.h>
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#include <dev/sound/pcm/ac97_patch.h>
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SND_DECLARE_FILE("$FreeBSD$");
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void ad1886_patch(struct ac97_info* codec)
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{
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#define AC97_AD_JACK_SPDIF 0x72
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/*
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* Presario700 workaround
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* for Jack Sense/SPDIF Register misetting causing
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* no audible output
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* by Santiago Nullo 04/05/2002
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*/
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ac97_wrcd(codec, AC97_AD_JACK_SPDIF, 0x0010);
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}
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void ad198x_patch(struct ac97_info* codec)
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{
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switch (ac97_getsubvendor(codec)) {
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case 0x11931043: /* Not for ASUS A9T (probably else too). */
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break;
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default:
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ac97_wrcd(codec, 0x76, ac97_rdcd(codec, 0x76) | 0x0420);
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break;
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}
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}
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void ad1981b_patch(struct ac97_info* codec)
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{
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/*
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* Enable headphone jack sensing.
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*/
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switch (ac97_getsubvendor(codec)) {
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case 0x02d91014: /* IBM Thinkcentre */
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case 0x099c103c: /* HP nx6110 */
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ac97_wrcd(codec, AC97_AD_JACK_SPDIF,
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ac97_rdcd(codec, AC97_AD_JACK_SPDIF) | 0x0800);
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break;
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default:
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break;
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}
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}
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void cmi9739_patch(struct ac97_info* codec)
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{
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/*
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* Few laptops need extra register initialization
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* to power up the internal speakers.
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*/
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switch (ac97_getsubvendor(codec)) {
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case 0x18431043: /* ASUS W1000N */
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ac97_wrcd(codec, AC97_REG_POWER, 0x000f);
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ac97_wrcd(codec, AC97_MIXEXT_CLFE, 0x0000);
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ac97_wrcd(codec, 0x64, 0x7110);
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break;
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default:
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break;
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}
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}
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void alc655_patch(struct ac97_info* codec)
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{
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/*
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* MSI (Micro-Star International) specific EAPD quirk.
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*/
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switch (ac97_getsubvendor(codec)) {
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case 0x00611462: /* MSI S250 */
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case 0x01311462: /* MSI S270 */
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case 0x01611462: /* LG K1 Express */
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case 0x03511462: /* MSI L725 */
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ac97_wrcd(codec, 0x7a, ac97_rdcd(codec, 0x7a) & 0xfffd);
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break;
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case 0x10ca1734:
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/*
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* Amilo Pro V2055 with ALC655 has phone out by default
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* disabled (surround on), leaving us only with internal
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* speakers. This should really go to mixer. We write the
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* Data Flow Control reg.
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*/
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ac97_wrcd(codec, 0x6a, ac97_rdcd(codec, 0x6a) | 0x0001);
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break;
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default:
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break;
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}
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}
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