freebsd-nq/sys/dev/sound/pcm/channel_if.m
Ariff Abdullah 90da2b2859 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

234 lines
5.4 KiB
Objective-C

#-
# KOBJ
#
# Copyright (c) 2005-2009 Ariff Abdullah <ariff@FreeBSD.org>
# Portions Copyright (c) Ryan Beasley <ryan.beasley@gmail.com> - GSoC 2006
# Copyright (c) 2000 Cameron Grant <cg@FreeBSD.org>
# All rights reserved.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions
# are met:
# 1. Redistributions of source code must retain the above copyright
# notice, this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright
# notice, this list of conditions and the following disclaimer in the
# documentation and/or other materials provided with the distribution.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
# ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
# ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
# FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
# DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
# OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
# HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
# LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
# OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
# SUCH DAMAGE.
#
# $FreeBSD$
#
#include <dev/sound/pcm/sound.h>
INTERFACE channel;
CODE {
static int
channel_noreset(kobj_t obj, void *data)
{
return 0;
}
static int
channel_noresetdone(kobj_t obj, void *data)
{
return 0;
}
static int
channel_nofree(kobj_t obj, void *data)
{
return 1;
}
static u_int32_t
channel_nogetptr(kobj_t obj, void *data)
{
return 0;
}
static int
channel_nonotify(kobj_t obj, void *data, u_int32_t changed)
{
return 0;
}
static int
channel_nogetpeaks(kobj_t obj, void *data, int *lpeak, int *rpeak)
{
return -1;
}
static int
channel_nogetrates(kobj_t obj, void *data, int **rates)
{
*rates = NULL;
return 0;
}
static int
channel_nosetfragments(kobj_t obj, void *data, u_int32_t blocksize, u_int32_t blockcount)
{
return ENOTSUP;
}
static struct pcmchan_matrix *
channel_nogetmatrix(kobj_t obj, void *data, u_int32_t format)
{
format = feeder_matrix_default_format(format);
return (feeder_matrix_format_map(format));
}
static int
channel_nosetmatrix(kobj_t obj, void *data, struct pcmchan_matrix *m)
{
return ENOTSUP;
}
};
METHOD void* init {
kobj_t obj;
void *devinfo;
struct snd_dbuf *b;
struct pcm_channel *c;
int dir;
};
METHOD int free {
kobj_t obj;
void *data;
} DEFAULT channel_nofree;
METHOD int reset {
kobj_t obj;
void *data;
} DEFAULT channel_noreset;
METHOD int resetdone {
kobj_t obj;
void *data;
} DEFAULT channel_noresetdone;
METHOD int setformat {
kobj_t obj;
void *data;
u_int32_t format;
};
METHOD u_int32_t setspeed {
kobj_t obj;
void *data;
u_int32_t speed;
};
METHOD u_int32_t setblocksize {
kobj_t obj;
void *data;
u_int32_t blocksize;
};
METHOD int setfragments {
kobj_t obj;
void *data;
u_int32_t blocksize;
u_int32_t blockcount;
} DEFAULT channel_nosetfragments;
METHOD int trigger {
kobj_t obj;
void *data;
int go;
};
METHOD u_int32_t getptr {
kobj_t obj;
void *data;
} DEFAULT channel_nogetptr;
METHOD struct pcmchan_caps* getcaps {
kobj_t obj;
void *data;
};
METHOD int notify {
kobj_t obj;
void *data;
u_int32_t changed;
} DEFAULT channel_nonotify;
/**
* @brief Retrieve channel peak values
*
* This function is intended to obtain peak volume values for samples
* played/recorded on a channel. Values are on a linear scale from 0 to
* 32767. If the channel is monaural, a single value should be recorded
* in @c lpeak.
*
* If hardware support isn't available, the SNDCTL_DSP_GET[IO]PEAKS
* operation should return EINVAL. However, we may opt to provide
* software support that the user may toggle via sysctl/mixext.
*
* @param obj standard kobj object (usually @c channel->methods)
* @param data driver-specific data (usually @c channel->devinfo)
* @param lpeak pointer to store left peak level
* @param rpeak pointer to store right peak level
*
* @retval -1 Error; usually operation isn't supported.
* @retval 0 success
*/
METHOD int getpeaks {
kobj_t obj;
void *data;
int *lpeak;
int *rpeak;
} DEFAULT channel_nogetpeaks;
/**
* @brief Retrieve discrete supported sample rates
*
* Some cards operate at fixed rates, and this call is intended to retrieve
* those rates primarily for when in-kernel rate adjustment is undesirable
* (e.g., application wants direct DMA access after setting a channel to run
* "uncooked").
*
* The parameter @c rates is a double pointer which will be reset to
* point to an array of supported sample rates. The number of elements
* in the array is returned to the caller.
*
* @param obj standard kobj object (usually @c channel->methods)
* @param data driver-specific data (usually @c channel->devinfo)
* @param rates rate array pointer
*
* @return Number of rates in the array
*/
METHOD int getrates {
kobj_t obj;
void *data;
int **rates;
} DEFAULT channel_nogetrates;
METHOD struct pcmchan_matrix * getmatrix {
kobj_t obj;
void *data;
u_int32_t format;
} DEFAULT channel_nogetmatrix;
METHOD int setmatrix {
kobj_t obj;
void *data;
struct pcmchan_matrix *m;
} DEFAULT channel_nosetmatrix;