audio/impl/audio_format.c

	audio_format.c revision 88447a05f537aabe9a1bc3d5313f22581ec992a7
/*
 * CDDL HEADER START
 *
 * The contents of this file are subject to the terms of the
 * Common Development and Distribution License (the "License").
 * You may not use this file except in compliance with the License.
 *
 * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 * or http://www.opensolaris.org/os/licensing.
 * See the License for the specific language governing permissions
 * and limitations under the License.
 *
 * When distributing Covered Code, include this CDDL HEADER in each
 * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
 * If applicable, add the following below this CDDL HEADER, with the
 * fields enclosed by brackets "[]" replaced with your own identifying
 * information: Portions Copyright [yyyy] [name of copyright owner]
 *
 * CDDL HEADER END
 */
/*
 * Copyright (C) 4Front Technologies 1996-2008.
 *
 * Copyright 2009 Sun Microsystems, Inc.  All rights reserved.
 * Use is subject to license terms.
 */

/*
 * Purpose: Audio format conversion routines used by audio.c
 */

#include <sys/types.h>
#include <sys/ddi.h>
#include <sys/sunddi.h>
#include <sys/audio/g711.h>

#include "audio_impl.h"
#include "audio_grc3.h"


/*
 * Note: In the function below, the division by the number of channels is
 * probably fairly expensive.  It turns out that we usually deal with stereo
 * or mono data, so perhaps it would be useful to build custom versions of
 * this function that only dealt with stereo or mono.
 */
static int
do_src(audio_stream_t *sp, void *p1, void *p2, int len, int nchan)
{
    int ch, size;

    /*
     * Note that we presume that we are doing sample rate
     * conversions on AUDIO_FORMAT_S24_NE, which means that have 4
     * byte and 32-bit samples.
     */
    size = sp->s_cnv_max / 4;       /* sample size is 4 */
    size /= nchan;

    for (ch = 0; ch < nchan; ch++) {
        grc3_convert(sp->s_src_state[ch], sp->s_src_quality,
            p1, p2, len, size, nchan, ch);
    }
    return (((grc3state_t *)sp->s_src_state[0])->outsz);
}

static void
setup_src(audio_stream_t *sp, int srate, int trate, int sch, int tch)
{
    int ch, nch;

    nch = min(sch, tch);

    ASSERT(nch <= AUDIO_MAX_CHANNELS);

    if (sp->s_src_quality < 1)
        sp->s_src_quality = 1;
    if (sp->s_src_quality > 5)
        sp->s_src_quality = 5;

    for (ch = 0; ch < nch; ch++) {
        grc3_reset(sp->s_src_state[ch]);
        grc3_setup(sp->s_src_state[ch], srate, trate);
    }
}

static int
cnv_srconly(audio_stream_t *sp, int len)
{
    void *src = sp->s_cnv_src;
    void *dst = sp->s_cnv_dst;

    /*
     * We must be using 24-bit native signed.
     */
    len = do_src(sp, src, dst, len, sp->s_cnv_src_nchan);

    sp->s_cnv_src = dst;
    sp->s_cnv_dst = src;

    return (len);
}

static int
cnv_s24oe(audio_stream_t *sp, int len)
{
    /*
     * Endian switch works in both directions.  We do it in place.
     */
    int32_t *src = sp->s_cnv_src;

    for (int i = len * sp->s_cnv_src_nchan; i; i--) {
        *src = ddi_swap32(*src);
        src++;
    }

    return (len);
}


static int
cnv_from_s8(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    int8_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (*src++) << 16;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_u8(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint8_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (int8_t)((*src++) ^ 0x80) << 16;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_ulaw(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint8_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--) {
        *dst++ = _8ulaw2linear16[(*src++)] << 8;
    }
    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_alaw(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint8_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--) {
        *dst++ = _8alaw2linear16[(*src++)] << 8;
    }
    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_s16ne(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    int16_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (*src++) << 8;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_s16oe(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    int16_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (int16_t)(ddi_swap16(*src++)) << 8;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_u16ne(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint16_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (int16_t)((*src++) ^ 0x8000) << 8;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_u16oe(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint16_t *src = s;
    int32_t *dst = d;

    for (int i = len * sp->s_cnv_src_nchan; i; i--)
        *dst++ = (int16_t)(ddi_swap16((*src++) ^ 0x8000)) << 8;

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_s24p(audio_stream_t *sp, int len)
{
    void *s = sp->s_cnv_src;
    void *d = sp->s_cnv_dst;
    uint8_t *src = s;
    int32_t *dst = d;
    int32_t tmp;

    for (int i = len * sp->s_cnv_src_nchan; i; i--) {
        /* NB: this is a little endian format */
        tmp = (*src++);
        tmp |= (*src++) << 8;
        tmp |= (*src++) << 16;
        *dst++ = tmp;
    }

    sp->s_cnv_src = d;
    sp->s_cnv_dst = s;
    return (len);
}

static int
cnv_from_s32ne(audio_stream_t *sp, int len)
{
    /* 32-bit conversions can be done in place */
    int32_t *src = sp->s_cnv_src;

    for (int i = len * sp->s_cnv_src_nchan; i; i--, src++)
        *src = *src >> 8;

    return (len);
}

static int
cnv_from_s32oe(audio_stream_t *sp, int len)
{
    /* 32-bit conversions can be done in place */
    int32_t *src = sp->s_cnv_src;

    for (int i = len * sp->s_cnv_src_nchan; i; i--, src++)
        *src = (int32_t)(ddi_swap32(*src)) >> 8;

    return (len);
}

/*
 * NB: All the destination format conversions use the same or fewer
 * bytes as the 24-bit unpacked (32-bits used per sample), so we can
 * convert all of them in place.
 */

static int
cnv_to_u8(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint8_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = (*src++ >> 16) ^ 0x80;

    return (len);
}

static int
cnv_to_s8(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    int8_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = *src++ >> 16;

    return (len);
}

static int
cnv_to_ulaw(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint8_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
        int idx = *src++;
        idx >>= 10;
        idx += G711_ULAW_MIDPOINT;
        idx &= 0x3fff;  /* safety precaution */
        *dst++ = _14linear2ulaw8[idx];
    }

    return (len);
}

static int
cnv_to_alaw(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint8_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
        int idx = *src++;
        idx >>= 11;
        idx += G711_ALAW_MIDPOINT;
        idx &= 0x1fff;  /* safety precaution */
        *dst++ = _13linear2alaw8[idx];
    }

    return (len);
}

static int
cnv_to_s16ne(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    int16_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = *src++ >> 8;

    return (len);
}

static int
cnv_to_s16oe(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    int16_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = ddi_swap16(*src++ >> 8);

    return (len);
}

static int
cnv_to_u16ne(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint16_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = (*src++ >> 8) ^ 0x8000;

    return (len);
}

static int
cnv_to_u16oe(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint16_t *dst = (void *)src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--)
        *dst++ = ddi_swap16(*src++ >> 8) ^ 0x8000;

    return (len);
}

static int
cnv_to_s24p(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;
    uint8_t *dst = (void *)src;
    int32_t d;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--) {
        /* NB: this is a little endian format */
        d = *src++;
        *dst++ = d & 0xff;
        *dst++ = (d >> 8) & 0xff;
        *dst++ = (d >> 16) & 0xff;
    }

    return (len);
}

static int
cnv_to_s32ne(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--, src++)
        *src = *src << 8;

    return (len);
}

static int
cnv_to_s32oe(audio_stream_t *sp, int len)
{
    int32_t *src = sp->s_cnv_src;

    for (int i = len * sp->s_cnv_dst_nchan; i; i--, src++)
        *src = ddi_swap32(*src << 8);

    return (len);
}

static int
cnv_default(audio_stream_t *sp, int len)
{
    /*
     * Note that the formats were already preverified during
     * select_converter, to ensure that only supported formats are
     * used.
     */

    /*
     * Convert samples to 24 bit (32 bit lsb aligned) if
     * necessary.
     */

    switch (sp->s_cnv_src_format) {

    case AUDIO_FORMAT_U8:
        len = cnv_from_u8(sp, len);
        break;

    case AUDIO_FORMAT_S8:
        len = cnv_from_s8(sp, len);
        break;

    case AUDIO_FORMAT_ULAW:
        len = cnv_from_ulaw(sp, len);
        break;

    case AUDIO_FORMAT_ALAW:
        len = cnv_from_alaw(sp, len);
        break;

    case AUDIO_FORMAT_S16_NE:
        len = cnv_from_s16ne(sp, len);
        break;

    case AUDIO_FORMAT_S16_OE:
        len = cnv_from_s16oe(sp, len);
        break;

    case AUDIO_FORMAT_U16_NE:
        len = cnv_from_u16ne(sp, len);
        break;

    case AUDIO_FORMAT_U16_OE:
        len = cnv_from_u16oe(sp, len);
        break;

    case AUDIO_FORMAT_S32_NE:
        len = cnv_from_s32ne(sp, len);
        break;

    case AUDIO_FORMAT_S32_OE:
        len = cnv_from_s32oe(sp, len);
        break;

    case AUDIO_FORMAT_S24_OE:
        len = cnv_s24oe(sp, len);
        break;

    case AUDIO_FORMAT_S24_PACKED:
        len = cnv_from_s24p(sp, len);
        break;
    }

    /*
     * If we aren't decreasing the number of channels, then do the
     * SRC now.  (We prefer to do SRC on the smaller number of channels.)
     */
    if (sp->s_cnv_src_rate != sp->s_cnv_dst_rate &&
        sp->s_cnv_src_nchan <= sp->s_cnv_dst_nchan) {
        int32_t *src = sp->s_cnv_src;
        int32_t *dst = sp->s_cnv_dst;

        len = do_src(sp, src, dst, len, sp->s_cnv_src_nchan);

        sp->s_cnv_src = dst;
        sp->s_cnv_dst = src;
    }

    /*
     * Convert between mono and stereo
     */

    if (sp->s_cnv_src_nchan != sp->s_cnv_dst_nchan) {
        int32_t *src = sp->s_cnv_src;
        int32_t *dst = sp->s_cnv_dst;
        int tc = sp->s_cnv_dst_nchan;
        int sc = sp->s_cnv_src_nchan;
        int nc;
        int i;

        sp->s_cnv_src = dst;
        sp->s_cnv_dst = src;

        if (sc == 1) {
            /*
             * Mono expansion.  We expand into the stereo
             * channel, and leave other channels silent.
             */
            for (i = len; i; i--) {
                *dst++ = *src;
                *dst++ = *src++;
                for (int j = tc - 2; j > 0; j--) {
                    *dst++ = 0;
                }

            }

        } else if (sc == 2 && tc == 1) {
            /*
             * Stereo -> mono.  We do stereo separately to make
             * the division fast (div by const 2 is just shift).
             */
            for (i = len; i; i--) {
                /*
                 * Take just the left channel sample,
                 * discard the right channel.
                 */
                *dst++ = *src++;    /* left */
                src++;          /* right */
            }
        } else {
            /*
             * Multi channel conversions.  We just copy the
             * minimum number of channels.
             */

            /* Calculate number of frames */

            nc = min(sc, tc);

            /* Clear destination */
            bzero(dst, (len * tc * sizeof (int32_t)));

            for (i = len; i; i--) {
                int c;

                for (c = 0; c < nc; c++)
                    dst[c] = src[c];

                src += sc;
                dst += tc;
            }
        }
    }

    /*
     * If we didn't do SRC pre-conversion, then do it now.
     */
    if (sp->s_cnv_src_rate != sp->s_cnv_dst_rate &&
        sp->s_cnv_src_nchan > sp->s_cnv_dst_nchan) {

        int32_t *src = sp->s_cnv_src;
        int32_t *dst = sp->s_cnv_dst;

        len = do_src(sp, src, dst, len, sp->s_cnv_dst_nchan);

        sp->s_cnv_src = dst;
        sp->s_cnv_dst = src;
    }

    /*
     * Finally convert samples from internal 24 bit format to target format
     */

    switch (sp->s_cnv_dst_format) {
    case AUDIO_FORMAT_U8:
        len = cnv_to_u8(sp, len);
        break;

    case AUDIO_FORMAT_S8:
        len = cnv_to_s8(sp, len);
        break;

    case AUDIO_FORMAT_S16_NE:
        len = cnv_to_s16ne(sp, len);
        break;

    case AUDIO_FORMAT_S16_OE:
        len = cnv_to_s16oe(sp, len);
        break;

    case AUDIO_FORMAT_U16_NE:
        len = cnv_to_u16ne(sp, len);
        break;

    case AUDIO_FORMAT_U16_OE:
        len = cnv_to_u16oe(sp, len);
        break;

    case AUDIO_FORMAT_S24_OE:
        len = cnv_s24oe(sp, len);
        break;

    case AUDIO_FORMAT_S24_PACKED:
        len = cnv_to_s24p(sp, len);
        break;

    case AUDIO_FORMAT_S32_NE:
        len = cnv_to_s32ne(sp, len);
        break;

    case AUDIO_FORMAT_S32_OE:
        len = cnv_to_s32oe(sp, len);
        break;

    case AUDIO_FORMAT_ULAW:
        len = cnv_to_ulaw(sp, len);
        break;

    case AUDIO_FORMAT_ALAW:
        len = cnv_to_alaw(sp, len);
        break;
    }

    return (len);
}

static const struct audio_format_info {
    unsigned        format;
    int         sampsize;
    audio_cnv_func_t    from;
    audio_cnv_func_t    to;
} audio_format_info[] = {
    { AUDIO_FORMAT_S8,  1,  cnv_from_s8,        cnv_to_s8 },
    { AUDIO_FORMAT_U8,  1,  cnv_from_u8,        cnv_to_u8 },
    { AUDIO_FORMAT_ULAW,    1,  cnv_from_ulaw,      cnv_to_ulaw },
    { AUDIO_FORMAT_ALAW,    1,  cnv_from_alaw,      cnv_to_alaw },
    { AUDIO_FORMAT_S16_NE,  2,  cnv_from_s16ne,     cnv_to_s16ne },
    { AUDIO_FORMAT_S16_OE,  2,  cnv_from_s16oe,     cnv_to_s16oe },
    { AUDIO_FORMAT_U16_NE,  2,  cnv_from_u16ne,     cnv_to_u16ne },
    { AUDIO_FORMAT_U16_OE,  2,  cnv_from_u16oe,     cnv_to_u16oe },
    { AUDIO_FORMAT_S32_NE,  4,  cnv_from_s32ne,     cnv_to_s32ne },
    { AUDIO_FORMAT_S32_OE,  4,  cnv_from_s32oe,     cnv_to_s32oe },

    /* 24-bit formats are "special" */
    { AUDIO_FORMAT_S24_NE,  4,  NULL,           NULL },
    { AUDIO_FORMAT_S24_OE,  4,  cnv_s24oe,      cnv_s24oe },
    { AUDIO_FORMAT_S24_PACKED, 3,   cnv_from_s24p,      cnv_to_s24p },

    /* sentinel */
    { AUDIO_FORMAT_NONE,    0,  NULL,           NULL }
};

int
auimpl_format_setup(audio_stream_t *sp, audio_parms_t *uparms)
{
    audio_parms_t           *source = &sp->s_cnv_src_parms;
    audio_parms_t           *target = &sp->s_cnv_dst_parms;
    audio_cnv_func_t        converter = NULL;
    const struct audio_format_info  *info;
    int             expand = AUDIO_UNIT_EXPAND;
    unsigned            cnv_sampsz = sizeof (uint32_t);
    unsigned            cnv_max;

    if (sp == &sp->s_client->c_ostream) {
        source = uparms;
    } else {
        target = uparms;
    }

    ASSERT(mutex_owned(&sp->s_lock));

    /*
     * At least one of the source or target are S24_NE.
     *
     * If we have a signed/native endian format, then pick an
     * optimized converter.  While at it, ensure that a valid
     * format is selected.
     */
    if (source->p_format != AUDIO_FORMAT_S24_NE) {
        for (info = &audio_format_info[0]; info->sampsize; info++) {
            if (source->p_format == info->format) {
                converter = info->from;
                expand *= sizeof (int32_t);
                expand /= info->sampsize;
                /* save source frame size */
                cnv_sampsz = info->sampsize;
                break;
            }
        }
        if (info->format == AUDIO_FORMAT_NONE) {
            audio_dev_warn(sp->s_client->c_dev,
                "invalid source format selected");
            return (EINVAL);
        }

    } else if (target->p_format != AUDIO_FORMAT_S24_NE) {
        for (info = &audio_format_info[0]; info->sampsize; info++) {
            if (target->p_format == info->format) {
                converter = info->to;
                expand *= info->sampsize;
                expand /= sizeof (int32_t);
                break;
            }
        }
        if (info->format == AUDIO_FORMAT_NONE) {
            audio_dev_warn(sp->s_client->c_dev,
                "invalid target format selected");
            return (EINVAL);
        }
    }

    if (source->p_nchan != target->p_nchan) {
        /*
         * if channels need conversion, then we must use the
         * default.
         */
        converter = cnv_default;
        expand *= target->p_nchan;
        expand /= source->p_nchan;
    }

    if (source->p_rate != target->p_rate) {
        /*
         * We need SRC; if we can avoid data conversion, do so.
         */
        setup_src(sp, source->p_rate, target->p_rate,
            source->p_nchan, target->p_nchan);

        converter = (converter == NULL) ? cnv_srconly : cnv_default;

        expand *= target->p_rate;
        expand /= source->p_rate;
    }

    ASSERT(sp->s_engine);
    ASSERT(sp->s_engine->e_intrs);

    /*
     * Figure out the size of the conversion buffer we need.  We
     * assume room for two full source fragments, which ought to
     * be enough, even with rounding errors.
     */
    cnv_max = 2 * (source->p_rate / sp->s_engine->e_intrs) *
        cnv_sampsz * source->p_nchan;

    /*
     * If the conversion will cause us to expand fragments, then
     * we need to increase cnv_max.  Scale by AUDIO_UNIT_EXPAND to
     * avoid rouding errors or losing bits when doing reducing
     * conversions.
     */
    if (expand > AUDIO_UNIT_EXPAND) {
        cnv_max *= expand;
        cnv_max /= AUDIO_UNIT_EXPAND;
    }

    /*
     * Now we need to allocate space.
     */
    if (sp->s_cnv_max < cnv_max) {
        uint32_t *buf0, *buf1;

        buf0 = kmem_alloc(cnv_max, KM_NOSLEEP);
        buf1 = kmem_alloc(cnv_max, KM_NOSLEEP);
        if ((buf0 == NULL) || (buf1 == NULL)) {
            audio_dev_warn(sp->s_client->c_dev,
                "failed to allocate audio conversion buffer "
                "(%u bytes)", cnv_max);
            if (buf0)
                kmem_free(buf0, cnv_max);
            if (buf1)
                kmem_free(buf1, cnv_max);
            return (ENOMEM);
        }

        if (sp->s_cnv_buf0)
            kmem_free(sp->s_cnv_buf0, sp->s_cnv_max);
        if (sp->s_cnv_buf1)
            kmem_free(sp->s_cnv_buf1, sp->s_cnv_max);

        sp->s_cnv_buf0 = buf0;
        sp->s_cnv_buf1 = buf1;
        sp->s_cnv_max = cnv_max;
    }

    /*
     * NB: From here on, we must not fail.
     */

    /*
     * Configure default fragment setup.
     */
    for (info = &audio_format_info[0]; info->sampsize; info++) {
        if (uparms->p_format == info->format) {
            break;
        }
    }
    ASSERT(info->sampsize);

    sp->s_framesz = info->sampsize * uparms->p_nchan;
    sp->s_fragfr = (uparms->p_rate / sp->s_engine->e_intrs);
    sp->s_fragbytes = sp->s_fragfr * sp->s_framesz;
    sp->s_nfrags = sp->s_allocsz / sp->s_fragbytes;
    sp->s_nframes = sp->s_nfrags * sp->s_fragfr;
    sp->s_nbytes = sp->s_nframes * sp->s_framesz;
    *sp->s_user_parms = *uparms;
    sp->s_converter = converter;

    /*
     * Ensure that we toss any stale data -- probably wrong format.
     * Note that as a consequence of this, all of the offsets and
     * counters get reset.  Clients should not rely on these values
     * being preserved when changing formats.
     *
     * Its critical that we reset the indices, in particular,
     * because not only will the data be the wrong format, but the
     * indices themselves are quite possibly going to be invalid.
     */
    sp->s_cnv_cnt = 0;
    sp->s_tail = sp->s_head = 0;
    sp->s_tidx = sp->s_hidx = 0;

    return (0);
}

int
auimpl_format_alloc(audio_stream_t *sp)
{
    int i;

    ASSERT(mutex_owned(&sp->s_lock));
    for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
        sp->s_src_state[i] =
            kmem_zalloc(sizeof (grc3state_t), KM_NOSLEEP);
        if (sp->s_src_state[i] == NULL) {
            audio_dev_warn(sp->s_client->c_dev,
                "unable to allocate SRC state structures");
            return (ENOMEM);
        }
    }
    return (0);
}

void
auimpl_format_free(audio_stream_t *sp)
{
    int i;

    for (i = 0; i < AUDIO_MAX_CHANNELS; i++) {
        if (sp->s_src_state[i] != NULL) {
            kmem_free(sp->s_src_state[i], sizeof (grc3state_t));
            sp->s_src_state[i] = NULL;
        }
    }
}