freebsd-dev/sys/dev/sound/pcm/feeder_rate.c
Ariff Abdullah a580b31a54 Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.

General
-------

- Multichannel safe, endian safe, format safe
   * Large part of critical pcm filters such as vchan.c, feeder_rate.c,
     feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
     using them does not cause the pcm data to be converted to 16bit little
     endian.
   * Macrosses for accessing pcm data safely are defined within sound.h in
     the form of PCM_READ_* / PCM_WRITE_*
   * Currently, most of them are probably limited for mono/stereo handling,
     but the future addition of true multichannel will be much easier.

- Low latency operation
  * Well, this require lot more works to do not just within sound driver,
    but we're heading towards right direction. Buffer/block sizing within
    channel.c is rewritten to calculate precise allocation for various
    combination of sample/data/rate size. As a result, applying correct
    SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
    to what commercial 4front driver do.
  * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
    result long delay.
  * Eliminate sound truncation if the sound data is too small.
    DIY:
      1) Download / extract
         http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
      2) Do a comparison between "cat state*.au > /dev/dsp" and
         "for x in state*.au ; do cat $x > /dev/dsp ; done"
         - there should be no "perceivable" differences.
    Double close for PR kern/31445.

  CAVEAT: Low latency come with (unbearable) price especially for poorly
          written applications. Applications that trying to act smarter
	  by requesting (wrong) blocksize/blockcount will suffer the most.
	  Fixup samples/patches can be found at:
	  http://people.freebsd.org/~ariff/ports/

- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
  due to closer compatibility with 4front driver.
  Discussed with: marcus@ (long time ago?)

- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
  moved to their own dev sysctl nodes, notably:
  hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
  Bump __FreeBSD_version.

Driver specific
---------------

- Ditto for sysctls.

- snd_atiixp, snd_es137x, snd_via8233, snd_hda
  * Numerous cleanups and fixes.
  * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
   This was intended for pure debugging and latency measurement, but proven
   good enough in few unexpected and rare cases (such as problematic shared
   IRQ with GIANT devices - USB). Polling can be enabled/disabled through
   dev.pcm.0.polling. Disabled by default.

- snd_ich
  * Fix possible overflow during speed calibration. Delay final
    initialization (pcm_setstatus) after calibration finished.
    PR: kern/100169
    Tested by: Kevin Overman <oberman@es.net>
  * Inverted EAPD for few Nec VersaPro.
    PR: kern/104715
    Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>

Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.

Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00

607 lines
19 KiB
C

/*-
* Copyright (c) 1999 Cameron Grant <cg@FreeBSD.org>
* Copyright (c) 2003 Orion Hodson <orion@FreeBSD.org>
* Copyright (c) 2005 Ariff Abdullah <ariff@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.
*
* 2006-02-21:
* ==========
*
* Major cleanup and overhaul to remove much redundant codes.
* Highlights:
* 1) Support for signed / unsigned 16, 24 and 32 bit,
* big / little endian,
* 2) Unlimited channels.
*
* 2005-06-11:
* ==========
*
* *New* and rewritten soft sample rate converter supporting arbitrary sample
* rates, fine grained scaling/coefficients and a unified up/down stereo
* converter. Most of the disclaimers from orion's notes also applies
* here, regarding linear interpolation deficiencies and pre/post
* anti-aliasing filtering issues. This version comes with a much simpler and
* tighter interface, although it works almost exactly like the older one.
*
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *
* *
* This new implementation is fully dedicated in memory of Cameron Grant, *
* the creator of the magnificent, highly addictive feeder infrastructure. *
* *
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *
*
* Orion's notes:
* =============
*
* This rate conversion code uses linear interpolation without any
* pre- or post- interpolation filtering to combat aliasing. This
* greatly limits the sound quality and should be addressed at some
* stage in the future.
*
* Since this accuracy of interpolation is sensitive and examination
* of the algorithm output is harder from the kernel, the code is
* designed to be compiled in the kernel and in a userland test
* harness. This is done by selectively including and excluding code
* with several portions based on whether _KERNEL is defined. It's a
* little ugly, but exceedingly useful. The testsuite and its
* revisions can be found at:
* http://people.freebsd.org/~orion/files/feedrate/
*
* Special thanks to Ken Marx for exposing flaws in the code and for
* testing revisions.
*/
#include <dev/sound/pcm/sound.h>
#include "feeder_if.h"
SND_DECLARE_FILE("$FreeBSD$");
#define RATE_ASSERT(x, y) /* KASSERT(x,y) */
#define RATE_TEST(x, y) /* if (!(x)) printf y */
#define RATE_TRACE(x...) /* printf(x) */
MALLOC_DEFINE(M_RATEFEEDER, "ratefeed", "pcm rate feeder");
/*
* Don't overflow 32bit integer, since everything is done
* within 32bit arithmetic.
*/
#define RATE_FACTOR_MIN 1
#define RATE_FACTOR_MAX PCM_S24_MAX
#define RATE_FACTOR_SAFE(val) (!((val) < RATE_FACTOR_MIN || \
(val) > RATE_FACTOR_MAX))
struct feed_rate_info;
typedef uint32_t (*feed_rate_converter)(struct feed_rate_info *, uint8_t *, uint32_t);
struct feed_rate_info {
uint32_t src, dst; /* rounded source / destination rates */
uint32_t rsrc, rdst; /* original source / destination rates */
uint32_t gx, gy; /* interpolation / decimation ratio */
uint32_t alpha; /* interpolation distance */
uint32_t pos, bpos; /* current sample / buffer positions */
uint32_t bufsz; /* total buffer size limit */
uint32_t bufsz_init; /* allocated buffer size */
uint32_t channels; /* total channels */
uint32_t bps; /* bytes-per-sample */
uint32_t stray; /* stray bytes */
uint8_t *buffer;
feed_rate_converter convert;
};
int feeder_rate_min = FEEDRATE_RATEMIN;
int feeder_rate_max = FEEDRATE_RATEMAX;
int feeder_rate_round = FEEDRATE_ROUNDHZ;
TUNABLE_INT("hw.snd.feeder_rate_min", &feeder_rate_min);
TUNABLE_INT("hw.snd.feeder_rate_max", &feeder_rate_max);
TUNABLE_INT("hw.snd.feeder_rate_round", &feeder_rate_round);
static int
sysctl_hw_snd_feeder_rate_min(SYSCTL_HANDLER_ARGS)
{
int err, val;
val = feeder_rate_min;
err = sysctl_handle_int(oidp, &val, sizeof(val), req);
if (RATE_FACTOR_SAFE(val) && val < feeder_rate_max)
feeder_rate_min = val;
else
err = EINVAL;
return err;
}
SYSCTL_PROC(_hw_snd, OID_AUTO, feeder_rate_min, CTLTYPE_INT | CTLFLAG_RW,
0, sizeof(int), sysctl_hw_snd_feeder_rate_min, "I",
"minimum allowable rate");
static int
sysctl_hw_snd_feeder_rate_max(SYSCTL_HANDLER_ARGS)
{
int err, val;
val = feeder_rate_max;
err = sysctl_handle_int(oidp, &val, sizeof(val), req);
if (RATE_FACTOR_SAFE(val) && val > feeder_rate_min)
feeder_rate_max = val;
else
err = EINVAL;
return err;
}
SYSCTL_PROC(_hw_snd, OID_AUTO, feeder_rate_max, CTLTYPE_INT | CTLFLAG_RW,
0, sizeof(int), sysctl_hw_snd_feeder_rate_max, "I",
"maximum allowable rate");
static int
sysctl_hw_snd_feeder_rate_round(SYSCTL_HANDLER_ARGS)
{
int err, val;
val = feeder_rate_round;
err = sysctl_handle_int(oidp, &val, sizeof(val), req);
if (val < FEEDRATE_ROUNDHZ_MIN || val > FEEDRATE_ROUNDHZ_MAX)
err = EINVAL;
else
feeder_rate_round = val - (val % FEEDRATE_ROUNDHZ);
return err;
}
SYSCTL_PROC(_hw_snd, OID_AUTO, feeder_rate_round, CTLTYPE_INT | CTLFLAG_RW,
0, sizeof(int), sysctl_hw_snd_feeder_rate_round, "I",
"sample rate converter rounding threshold");
#define FEEDER_RATE_CONVERT(FMTBIT, RATE_INTCAST, SIGN, SIGNS, ENDIAN, ENDIANS) \
static uint32_t \
feed_convert_##SIGNS##FMTBIT##ENDIANS(struct feed_rate_info *info, \
uint8_t *dst, uint32_t max) \
{ \
uint32_t ret, smpsz, bps, ch, pos, bpos, gx, gy, alpha, distance; \
int32_t x, y; \
int i; \
uint8_t *src, *sx, *sy; \
\
ret = 0; \
alpha = info->alpha; \
gx = info->gx; \
gy = info->gy; \
pos = info->pos; \
bpos = info->bpos; \
src = info->buffer + pos; \
ch = info->channels; \
bps = info->bps; \
smpsz = bps * ch; \
for (;;) { \
if (alpha < gx) { \
alpha += gy; \
pos += smpsz; \
if (pos == bpos) \
break; \
src += smpsz; \
} else { \
alpha -= gx; \
distance = (alpha << PCM_FXSHIFT) / gy; \
sx = src - smpsz; \
sy = src; \
i = ch; \
do { \
x = PCM_READ_##SIGN##FMTBIT##_##ENDIAN(sx); \
y = PCM_READ_##SIGN##FMTBIT##_##ENDIAN(sy); \
x = (((RATE_INTCAST)x * distance) + \
((RATE_INTCAST)y * ((1 << PCM_FXSHIFT) - \
distance))) >> PCM_FXSHIFT; \
PCM_WRITE_##SIGN##FMTBIT##_##ENDIAN(dst, x); \
dst += bps; \
sx += bps; \
sy += bps; \
ret += bps; \
} while (--i); \
if (ret == max) \
break; \
} \
} \
info->alpha = alpha; \
info->pos = pos; \
return ret; \
}
FEEDER_RATE_CONVERT(8, int32_t, S, s, NE, ne)
FEEDER_RATE_CONVERT(16, int32_t, S, s, LE, le)
FEEDER_RATE_CONVERT(24, int32_t, S, s, LE, le)
FEEDER_RATE_CONVERT(32, intpcm_t, S, s, LE, le)
FEEDER_RATE_CONVERT(16, int32_t, S, s, BE, be)
FEEDER_RATE_CONVERT(24, int32_t, S, s, BE, be)
FEEDER_RATE_CONVERT(32, intpcm_t, S, s, BE, be)
/* unsigned */
FEEDER_RATE_CONVERT(8, int32_t, U, u, NE, ne)
FEEDER_RATE_CONVERT(16, int32_t, U, u, LE, le)
FEEDER_RATE_CONVERT(24, int32_t, U, u, LE, le)
FEEDER_RATE_CONVERT(32, intpcm_t, U, u, LE, le)
FEEDER_RATE_CONVERT(16, int32_t, U, u, BE, be)
FEEDER_RATE_CONVERT(24, int32_t, U, u, BE, be)
FEEDER_RATE_CONVERT(32, intpcm_t, U, u, BE, be)
static void
feed_speed_ratio(uint32_t src, uint32_t dst, uint32_t *gx, uint32_t *gy)
{
uint32_t w, x = src, y = dst;
while (y != 0) {
w = x % y;
x = y;
y = w;
}
*gx = src / x;
*gy = dst / x;
}
static void
feed_rate_reset(struct feed_rate_info *info)
{
info->src = info->rsrc - (info->rsrc %
((feeder_rate_round > 0) ? feeder_rate_round : 1));
info->dst = info->rdst - (info->rdst %
((feeder_rate_round > 0) ? feeder_rate_round : 1));
info->gx = 1;
info->gy = 1;
info->alpha = 0;
info->channels = 2;
info->bps = 2;
info->convert = NULL;
info->bufsz = info->bufsz_init;
info->pos = 4;
info->bpos = 8;
info->stray = 0;
}
static int
feed_rate_setup(struct pcm_feeder *f)
{
struct feed_rate_info *info = f->data;
static const struct {
uint32_t format; /* pcm / audio format */
uint32_t bps; /* bytes-per-sample, regardless of
total channels */
feed_rate_converter convert;
} convtbl[] = {
{ AFMT_S8, PCM_8_BPS, feed_convert_s8ne },
{ AFMT_S16_LE, PCM_16_BPS, feed_convert_s16le },
{ AFMT_S24_LE, PCM_24_BPS, feed_convert_s24le },
{ AFMT_S32_LE, PCM_32_BPS, feed_convert_s32le },
{ AFMT_S16_BE, PCM_16_BPS, feed_convert_s16be },
{ AFMT_S24_BE, PCM_24_BPS, feed_convert_s24be },
{ AFMT_S32_BE, PCM_32_BPS, feed_convert_s32be },
/* unsigned */
{ AFMT_U8, PCM_8_BPS, feed_convert_u8ne },
{ AFMT_U16_LE, PCM_16_BPS, feed_convert_u16le },
{ AFMT_U24_LE, PCM_24_BPS, feed_convert_u24le },
{ AFMT_U32_LE, PCM_32_BPS, feed_convert_u32le },
{ AFMT_U16_BE, PCM_16_BPS, feed_convert_u16be },
{ AFMT_U24_BE, PCM_24_BPS, feed_convert_u24be },
{ AFMT_U32_BE, PCM_32_BPS, feed_convert_u32be },
{ 0, 0, NULL },
};
uint32_t i;
feed_rate_reset(info);
if (info->src != info->dst)
feed_speed_ratio(info->src, info->dst,
&info->gx, &info->gy);
if (!(RATE_FACTOR_SAFE(info->gx) && RATE_FACTOR_SAFE(info->gy)))
return -1;
for (i = 0; i < sizeof(convtbl) / sizeof(*convtbl); i++) {
if (convtbl[i].format == 0)
return -1;
if ((f->desc->out & ~AFMT_STEREO) == convtbl[i].format) {
info->bps = convtbl[i].bps;
info->convert = convtbl[i].convert;
break;
}
}
/*
* No need to interpolate/decimate, just do plain copy.
*/
if (info->gx == info->gy)
info->convert = NULL;
info->channels = (f->desc->out & AFMT_STEREO) ? 2 : 1;
info->pos = info->bps * info->channels;
info->bpos = info->pos << 1;
info->bufsz -= info->bufsz % info->pos;
memset(info->buffer, sndbuf_zerodata(f->desc->out), info->bpos);
RATE_TRACE("%s: %u (%u) -> %u (%u) [%u/%u] , "
"format=0x%08x, channels=%u, bufsz=%u\n",
__func__, info->src, info->rsrc, info->dst, info->rdst,
info->gx, info->gy,
f->desc->out, info->channels,
info->bufsz - info->pos);
return 0;
}
static int
feed_rate_set(struct pcm_feeder *f, int what, int32_t value)
{
struct feed_rate_info *info = f->data;
if (value < feeder_rate_min || value > feeder_rate_max)
return -1;
switch (what) {
case FEEDRATE_SRC:
info->rsrc = value;
break;
case FEEDRATE_DST:
info->rdst = value;
break;
default:
return -1;
}
return feed_rate_setup(f);
}
static int
feed_rate_get(struct pcm_feeder *f, int what)
{
struct feed_rate_info *info = f->data;
switch (what) {
case FEEDRATE_SRC:
return info->rsrc;
case FEEDRATE_DST:
return info->rdst;
default:
return -1;
}
return -1;
}
static int
feed_rate_init(struct pcm_feeder *f)
{
struct feed_rate_info *info;
if (f->desc->out != f->desc->in)
return EINVAL;
info = malloc(sizeof(*info), M_RATEFEEDER, M_NOWAIT | M_ZERO);
if (info == NULL)
return ENOMEM;
/*
* bufsz = sample from last cycle + conversion space
*/
info->bufsz_init = 8 + feeder_buffersize;
info->buffer = malloc(sizeof(*info->buffer) * info->bufsz_init,
M_RATEFEEDER, M_NOWAIT | M_ZERO);
if (info->buffer == NULL) {
free(info, M_RATEFEEDER);
return ENOMEM;
}
info->rsrc = DSP_DEFAULT_SPEED;
info->rdst = DSP_DEFAULT_SPEED;
f->data = info;
return feed_rate_setup(f);
}
static int
feed_rate_free(struct pcm_feeder *f)
{
struct feed_rate_info *info = f->data;
if (info) {
if (info->buffer)
free(info->buffer, M_RATEFEEDER);
free(info, M_RATEFEEDER);
}
f->data = NULL;
return 0;
}
static int
feed_rate(struct pcm_feeder *f, struct pcm_channel *c, uint8_t *b,
uint32_t count, void *source)
{
struct feed_rate_info *info = f->data;
uint32_t i, smpsz;
int32_t fetch, slot;
if (info->convert == NULL)
return FEEDER_FEED(f->source, c, b, count, source);
/*
* This loop has been optimized to generalize both up / down
* sampling without causing missing samples or excessive buffer
* feeding. The tricky part is to calculate *precise* (slot) value
* needed for the entire conversion space since we are bound to
* return and fill up the buffer according to the requested 'count'.
* Too much feeding will cause the extra buffer stay within temporary
* circular buffer forever and always manifest itself as a truncated
* sound during end of playback / recording. Too few, and we end up
* with possible underruns and waste of cpu cycles.
*
* 'Stray' management exist to combat with possible unaligned
* buffering by the caller.
*/
smpsz = info->bps * info->channels;
RATE_TEST(count >= smpsz && (count % smpsz) == 0,
("%s: Count size not sample integral (%d)\n", __func__, count));
if (count < smpsz)
return 0;
count -= count % smpsz;
/*
* This slot count formula will stay here for the next million years
* to come. This is the key of our circular buffering precision.
*/
slot = (((info->gx * (count / smpsz)) + info->gy - info->alpha - 1) / info->gy) * smpsz;
RATE_TEST((slot % smpsz) == 0, ("%s: Slot count not sample integral (%d)\n",
__func__, slot));
RATE_TEST(info->stray == 0, ("%s: [1] Stray bytes: %u\n",
__func__,info->stray));
if (info->pos != smpsz && info->bpos - info->pos == smpsz &&
info->bpos + slot > info->bufsz) {
/*
* Copy last unit sample and its previous to
* beginning of buffer.
*/
bcopy(info->buffer + info->pos - smpsz, info->buffer,
sizeof(*info->buffer) * (smpsz << 1));
info->pos = smpsz;
info->bpos = smpsz << 1;
}
RATE_ASSERT(slot >= 0, ("%s: Negative Slot: %d\n",
__func__, slot));
i = 0;
for (;;) {
for (;;) {
fetch = info->bufsz - info->bpos;
fetch -= info->stray;
RATE_ASSERT(fetch >= 0,
("%s: [1] Buffer overrun: %d > %d\n",
__func__, info->bpos, info->bufsz));
if (slot < fetch)
fetch = slot;
if (fetch > 0) {
RATE_ASSERT((int32_t)(info->bpos - info->stray) >= 0 &&
(info->bpos - info->stray) < info->bufsz,
("%s: DANGER - BUFFER OVERRUN! bufsz=%d, pos=%d\n", __func__,
info->bufsz, info->bpos - info->stray));
fetch = FEEDER_FEED(f->source, c,
info->buffer + info->bpos - info->stray,
fetch, source);
info->stray = 0;
if (fetch == 0)
break;
RATE_TEST((fetch % smpsz) == 0,
("%s: Fetch size not sample integral (%d)\n",
__func__, fetch));
info->stray += fetch % smpsz;
RATE_TEST(info->stray == 0,
("%s: Stray bytes detected (%d)\n",
__func__, info->stray));
fetch -= fetch % smpsz;
info->bpos += fetch;
slot -= fetch;
RATE_ASSERT(slot >= 0,
("%s: Negative Slot: %d\n", __func__,
slot));
if (slot == 0)
break;
if (info->bpos == info->bufsz)
break;
} else
break;
}
if (info->pos == info->bpos) {
RATE_TEST(info->pos == smpsz,
("%s: EOF while in progress\n", __func__));
break;
}
RATE_ASSERT(info->pos <= info->bpos,
("%s: [2] Buffer overrun: %d > %d\n", __func__,
info->pos, info->bpos));
RATE_ASSERT(info->pos < info->bpos,
("%s: Zero buffer!\n", __func__));
RATE_ASSERT(((info->bpos - info->pos) % smpsz) == 0,
("%s: Buffer not sample integral (%d)\n",
__func__, info->bpos - info->pos));
i += info->convert(info, b + i, count - i);
RATE_ASSERT(info->pos <= info->bpos,
("%s: [3] Buffer overrun: %d > %d\n",
__func__, info->pos, info->bpos));
if (info->pos == info->bpos) {
/*
* End of buffer cycle. Copy last unit sample
* to beginning of buffer so next cycle can
* interpolate using it.
*/
RATE_TEST(info->stray == 0, ("%s: [2] Stray bytes: %u\n", __func__, info->stray));
bcopy(info->buffer + info->pos - smpsz, info->buffer,
sizeof(*info->buffer) * smpsz);
info->bpos = smpsz;
info->pos = smpsz;
}
if (i == count)
break;
}
RATE_TEST((slot == 0 && count == i) ||
(slot > 0 && count > i &&
info->pos == info->bpos && info->pos == smpsz),
("%s: Inconsistent slot/count! "
"Count Expect: %u , Got: %u, Slot Left: %d\n",
__func__, count, i, slot));
RATE_TEST(info->stray == 0, ("%s: [3] Stray bytes: %u\n", __func__, info->stray));
return i;
}
static struct pcm_feederdesc feeder_rate_desc[] = {
{FEEDER_RATE, AFMT_S8, AFMT_S8, 0},
{FEEDER_RATE, AFMT_S16_LE, AFMT_S16_LE, 0},
{FEEDER_RATE, AFMT_S24_LE, AFMT_S24_LE, 0},
{FEEDER_RATE, AFMT_S32_LE, AFMT_S32_LE, 0},
{FEEDER_RATE, AFMT_S16_BE, AFMT_S16_BE, 0},
{FEEDER_RATE, AFMT_S24_BE, AFMT_S24_BE, 0},
{FEEDER_RATE, AFMT_S32_BE, AFMT_S32_BE, 0},
{FEEDER_RATE, AFMT_S8 | AFMT_STEREO, AFMT_S8 | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S16_LE | AFMT_STEREO, AFMT_S16_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S24_LE | AFMT_STEREO, AFMT_S24_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S32_LE | AFMT_STEREO, AFMT_S32_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S16_BE | AFMT_STEREO, AFMT_S16_BE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S24_BE | AFMT_STEREO, AFMT_S24_BE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_S32_BE | AFMT_STEREO, AFMT_S32_BE | AFMT_STEREO, 0},
/* unsigned */
{FEEDER_RATE, AFMT_U8, AFMT_U8, 0},
{FEEDER_RATE, AFMT_U16_LE, AFMT_U16_LE, 0},
{FEEDER_RATE, AFMT_U24_LE, AFMT_U24_LE, 0},
{FEEDER_RATE, AFMT_U32_LE, AFMT_U32_LE, 0},
{FEEDER_RATE, AFMT_U16_BE, AFMT_U16_BE, 0},
{FEEDER_RATE, AFMT_U24_BE, AFMT_U24_BE, 0},
{FEEDER_RATE, AFMT_U32_BE, AFMT_U32_BE, 0},
{FEEDER_RATE, AFMT_U8 | AFMT_STEREO, AFMT_U8 | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U16_LE | AFMT_STEREO, AFMT_U16_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U24_LE | AFMT_STEREO, AFMT_U24_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U32_LE | AFMT_STEREO, AFMT_U32_LE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U16_BE | AFMT_STEREO, AFMT_U16_BE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U24_BE | AFMT_STEREO, AFMT_U24_BE | AFMT_STEREO, 0},
{FEEDER_RATE, AFMT_U32_BE | AFMT_STEREO, AFMT_U32_BE | AFMT_STEREO, 0},
{0, 0, 0, 0},
};
static kobj_method_t feeder_rate_methods[] = {
KOBJMETHOD(feeder_init, feed_rate_init),
KOBJMETHOD(feeder_free, feed_rate_free),
KOBJMETHOD(feeder_set, feed_rate_set),
KOBJMETHOD(feeder_get, feed_rate_get),
KOBJMETHOD(feeder_feed, feed_rate),
{0, 0}
};
FEEDER_DECLARE(feeder_rate, 2, NULL);