path: root/audio/alsaaudio.c

/*
 * QEMU ALSA audio driver
 *
 * Copyright (c) 2005 Vassili Karpov (malc)
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
#include <alsa/asoundlib.h>
#include "qemu-common.h"
#include "audio.h"

#if QEMU_GNUC_PREREQ(4, 3)
#pragma GCC diagnostic ignored "-Waddress"
#endif

#define AUDIO_CAP "alsa"
#include "audio_int.h"

typedef struct ALSAVoiceOut {
    HWVoiceOut hw;
    void *pcm_buf;
    snd_pcm_t *handle;
} ALSAVoiceOut;

typedef struct ALSAVoiceIn {
    HWVoiceIn hw;
    snd_pcm_t *handle;
    void *pcm_buf;
} ALSAVoiceIn;

static struct {
    int size_in_usec_in;
    int size_in_usec_out;
    const char *pcm_name_in;
    const char *pcm_name_out;
    unsigned int buffer_size_in;
    unsigned int period_size_in;
    unsigned int buffer_size_out;
    unsigned int period_size_out;
    unsigned int threshold;

    int buffer_size_in_overridden;
    int period_size_in_overridden;

    int buffer_size_out_overridden;
    int period_size_out_overridden;
    int verbose;
} conf = {
    .buffer_size_out = 1024,
    .pcm_name_out = "default",
    .pcm_name_in = "default",
};

struct alsa_params_req {
    int freq;
    snd_pcm_format_t fmt;
    int nchannels;
    int size_in_usec;
    int override_mask;
    unsigned int buffer_size;
    unsigned int period_size;
};

struct alsa_params_obt {
    int freq;
    audfmt_e fmt;
    int endianness;
    int nchannels;
    snd_pcm_uframes_t samples;
};

static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
{
    va_list ap;

    va_start (ap, fmt);
    AUD_vlog (AUDIO_CAP, fmt, ap);
    va_end (ap);

    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}

static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
    int err,
    const char *typ,
    const char *fmt,
    ...
    )
{
    va_list ap;

    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);

    va_start (ap, fmt);
    AUD_vlog (AUDIO_CAP, fmt, ap);
    va_end (ap);

    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}

static void alsa_anal_close (snd_pcm_t **handlep)
{
    int err = snd_pcm_close (*handlep);
    if (err) {
        alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
    }
    *handlep = NULL;
}

static int alsa_write (SWVoiceOut *sw, void *buf, int len)
{
    return audio_pcm_sw_write (sw, buf, len);
}

static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
{
    switch (fmt) {
    case AUD_FMT_S8:
        return SND_PCM_FORMAT_S8;

    case AUD_FMT_U8:
        return SND_PCM_FORMAT_U8;

    case AUD_FMT_S16:
        return SND_PCM_FORMAT_S16_LE;

    case AUD_FMT_U16:
        return SND_PCM_FORMAT_U16_LE;

    case AUD_FMT_S32:
        return SND_PCM_FORMAT_S32_LE;

    case AUD_FMT_U32:
        return SND_PCM_FORMAT_U32_LE;

    default:
        dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
        abort ();
#endif
        return SND_PCM_FORMAT_U8;
    }
}

static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
                           int *endianness)
{
    switch (alsafmt) {
    case SND_PCM_FORMAT_S8:
        *endianness = 0;
        *fmt = AUD_FMT_S8;
        break;

    case SND_PCM_FORMAT_U8:
        *endianness = 0;
        *fmt = AUD_FMT_U8;
        break;

    case SND_PCM_FORMAT_S16_LE:
        *endianness = 0;
        *fmt = AUD_FMT_S16;
        break;

    case SND_PCM_FORMAT_U16_LE:
        *endianness = 0;
        *fmt = AUD_FMT_U16;
        break;

    case SND_PCM_FORMAT_S16_BE:
        *endianness = 1;
        *fmt = AUD_FMT_S16;
        break;

    case SND_PCM_FORMAT_U16_BE:
        *endianness = 1;
        *fmt = AUD_FMT_U16;
        break;

    case SND_PCM_FORMAT_S32_LE:
        *endianness = 0;
        *fmt = AUD_FMT_S32;
        break;

    case SND_PCM_FORMAT_U32_LE:
        *endianness = 0;
        *fmt = AUD_FMT_U32;
        break;

    case SND_PCM_FORMAT_S32_BE:
        *endianness = 1;
        *fmt = AUD_FMT_S32;
        break;

    case SND_PCM_FORMAT_U32_BE:
        *endianness = 1;
        *fmt = AUD_FMT_U32;
        break;

    default:
        dolog ("Unrecognized audio format %d\n", alsafmt);
        return -1;
    }

    return 0;
}

static void alsa_dump_info (struct alsa_params_req *req,
                            struct alsa_params_obt *obt)
{
    dolog ("parameter | requested value | obtained value\n");
    dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
    dolog ("channels  |      %10d |     %10d\n",
           req->nchannels, obt->nchannels);
    dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
    dolog ("============================================\n");
    dolog ("requested: buffer size %d period size %d\n",
           req->buffer_size, req->period_size);
    dolog ("obtained: samples %ld\n", obt->samples);
}

static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
    int err;
    snd_pcm_sw_params_t *sw_params;

    snd_pcm_sw_params_alloca (&sw_params);

    err = snd_pcm_sw_params_current (handle, sw_params);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to get current software parameters\n");
        return;
    }

    err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to set software threshold to %ld\n",
                     threshold);
        return;
    }

    err = snd_pcm_sw_params (handle, sw_params);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to set software parameters\n");
        return;
    }
}

static int alsa_open (int in, struct alsa_params_req *req,
                      struct alsa_params_obt *obt, snd_pcm_t **handlep)
{
    snd_pcm_t *handle;
    snd_pcm_hw_params_t *hw_params;
    int err;
    int size_in_usec;
    unsigned int freq, nchannels;
    const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
    snd_pcm_uframes_t obt_buffer_size;
    const char *typ = in ? "ADC" : "DAC";
    snd_pcm_format_t obtfmt;

    freq = req->freq;
    nchannels = req->nchannels;
    size_in_usec = req->size_in_usec;

    snd_pcm_hw_params_alloca (&hw_params);

    err = snd_pcm_open (
        &handle,
        pcm_name,
        in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
        SND_PCM_NONBLOCK
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
        return -1;
    }

    err = snd_pcm_hw_params_any (handle, hw_params);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
        goto err;
    }

    err = snd_pcm_hw_params_set_access (
        handle,
        hw_params,
        SND_PCM_ACCESS_RW_INTERLEAVED
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set access type\n");
        goto err;
    }

    err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
    if (err < 0 && conf.verbose) {
        alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
    }

    err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
        goto err;
    }

    err = snd_pcm_hw_params_set_channels_near (
        handle,
        hw_params,
        &nchannels
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
                      req->nchannels);
        goto err;
    }

    if (nchannels != 1 && nchannels != 2) {
        alsa_logerr2 (err, typ,
                      "Can not handle obtained number of channels %d\n",
                      nchannels);
        goto err;
    }

    if (req->buffer_size) {
        unsigned long obt;

        if (size_in_usec) {
            int dir = 0;
            unsigned int btime = req->buffer_size;

            err = snd_pcm_hw_params_set_buffer_time_near (
                handle,
                hw_params,
                &btime,
                &dir
                );
            obt = btime;
        }
        else {
            snd_pcm_uframes_t bsize = req->buffer_size;

            err = snd_pcm_hw_params_set_buffer_size_near (
                handle,
                hw_params,
                &bsize
                );
            obt = bsize;
        }
        if (err < 0) {
            alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
                          size_in_usec ? "time" : "size", req->buffer_size);
            goto err;
        }

        if ((req->override_mask & 2) && (obt - req->buffer_size))
            dolog ("Requested buffer %s %u was rejected, using %lu\n",
                   size_in_usec ? "time" : "size", req->buffer_size, obt);
    }

    if (req->period_size) {
        unsigned long obt;

        if (size_in_usec) {
            int dir = 0;
            unsigned int ptime = req->period_size;

            err = snd_pcm_hw_params_set_period_time_near (
                handle,
                hw_params,
                &ptime,
                &dir
                );
            obt = ptime;
        }
        else {
            int dir = 0;
            snd_pcm_uframes_t psize = req->period_size;

            err = snd_pcm_hw_params_set_period_size_near (
                handle,
                hw_params,
                &psize,
                &dir
                );
            obt = psize;
        }

        if (err < 0) {
            alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
                          size_in_usec ? "time" : "size", req->period_size);
            goto err;
        }

        if ((req->override_mask & 1) && (obt - req->period_size))
            dolog ("Requested period %s %u was rejected, using %lu\n",
                   size_in_usec ? "time" : "size", req->period_size, obt);
    }

    err = snd_pcm_hw_params (handle, hw_params);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
        goto err;
    }

    err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to get buffer size\n");
        goto err;
    }

    err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to get format\n");
        goto err;
    }

    if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
        dolog ("Invalid format was returned %d\n", obtfmt);
        goto err;
    }

    err = snd_pcm_prepare (handle);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
        goto err;
    }

    if (!in && conf.threshold) {
        snd_pcm_uframes_t threshold;
        int bytes_per_sec;

        bytes_per_sec = freq << (nchannels == 2);

        switch (obt->fmt) {
        case AUD_FMT_S8:
        case AUD_FMT_U8:
            break;

        case AUD_FMT_S16:
        case AUD_FMT_U16:
            bytes_per_sec <<= 1;
            break;

        case AUD_FMT_S32:
        case AUD_FMT_U32:
            bytes_per_sec <<= 2;
            break;
        }

        threshold = (conf.threshold * bytes_per_sec) / 1000;
        alsa_set_threshold (handle, threshold);
    }

    obt->nchannels = nchannels;
    obt->freq = freq;
    obt->samples = obt_buffer_size;

    *handlep = handle;

    if (conf.verbose &&
        (obt->fmt != req->fmt ||
         obt->nchannels != req->nchannels ||
         obt->freq != req->freq)) {
        dolog ("Audio paramters for %s\n", typ);
        alsa_dump_info (req, obt);
    }

#ifdef DEBUG
    alsa_dump_info (req, obt);
#endif
    return 0;

 err:
    alsa_anal_close (&handle);
    return -1;
}

static int alsa_recover (snd_pcm_t *handle)
{
    int err = snd_pcm_prepare (handle);
    if (err < 0) {
        alsa_logerr (err, "Failed to prepare handle %p\n", handle);
        return -1;
    }
    return 0;
}

static int alsa_resume (snd_pcm_t *handle)
{
    int err = snd_pcm_resume (handle);
    if (err < 0) {
        alsa_logerr (err, "Failed to resume handle %p\n", handle);
        return -1;
    }
    return 0;
}

static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
{
    snd_pcm_sframes_t avail;

    avail = snd_pcm_avail_update (handle);
    if (avail < 0) {
        if (avail == -EPIPE) {
            if (!alsa_recover (handle)) {
                avail = snd_pcm_avail_update (handle);
            }
        }

        if (avail < 0) {
            alsa_logerr (avail,
                         "Could not obtain number of available frames\n");
            return -1;
        }
    }

    return avail;
}

static int alsa_run_out (HWVoiceOut *hw)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
    int rpos, live, decr;
    int samples;
    uint8_t *dst;
    struct st_sample *src;
    snd_pcm_sframes_t avail;

    live = audio_pcm_hw_get_live_out (hw);
    if (!live) {
        return 0;
    }

    avail = alsa_get_avail (alsa->handle);
    if (avail < 0) {
        dolog ("Could not get number of available playback frames\n");
        return 0;
    }

    decr = audio_MIN (live, avail);
    samples = decr;
    rpos = hw->rpos;
    while (samples) {
        int left_till_end_samples = hw->samples - rpos;
        int len = audio_MIN (samples, left_till_end_samples);
        snd_pcm_sframes_t written;

        src = hw->mix_buf + rpos;
        dst = advance (alsa->pcm_buf, rpos << hw->info.shift);

        hw->clip (dst, src, len);

        while (len) {
            written = snd_pcm_writei (alsa->handle, dst, len);

            if (written <= 0) {
                switch (written) {
                case 0:
                    if (conf.verbose) {
                        dolog ("Failed to write %d frames (wrote zero)\n", len);
                    }
                    goto exit;

                case -EPIPE:
                    if (alsa_recover (alsa->handle)) {
                        alsa_logerr (written, "Failed to write %d frames\n",
                                     len);
                        goto exit;
                    }
                    if (conf.verbose) {
                        dolog ("Recovering from playback xrun\n");
                    }
                    continue;

                case -ESTRPIPE:
                    /* stream is suspended and waiting for an
                       application recovery */
                    if (alsa_resume (alsa->handle)) {
                        alsa_logerr (written, "Failed to write %d frames\n",
                                     len);
                        goto exit;
                    }
                    if (conf.verbose) {
                        dolog ("Resuming suspended output stream\n");
                    }
                    continue;

                case -EAGAIN:
                    goto exit;

                default:
                    alsa_logerr (written, "Failed to write %d frames to %p\n",
                                 len, dst);
                    goto exit;
                }
            }

            rpos = (rpos + written) % hw->samples;
            samples -= written;
            len -= written;
            dst = advance (dst, written << hw->info.shift);
            src += written;
        }
    }

 exit:
    hw->rpos = rpos;
    return decr;
}

static void alsa_fini_out (HWVoiceOut *hw)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    ldebug ("alsa_fini\n");
    alsa_anal_close (&alsa->handle);

    if (alsa->pcm_buf) {
        qemu_free (alsa->pcm_buf);
        alsa->pcm_buf = NULL;
    }
}

static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
    struct alsa_params_req req;
    struct alsa_params_obt obt;
    snd_pcm_t *handle;
    struct audsettings obt_as;

    req.fmt = aud_to_alsafmt (as->fmt);
    req.freq = as->freq;
    req.nchannels = as->nchannels;
    req.period_size = conf.period_size_out;
    req.buffer_size = conf.buffer_size_out;
    req.size_in_usec = conf.size_in_usec_out;
    req.override_mask =
        (conf.period_size_out_overridden ? 1 : 0) |
        (conf.buffer_size_out_overridden ? 2 : 0);

    if (alsa_open (0, &req, &obt, &handle)) {
        return -1;
    }

    obt_as.freq = obt.freq;
    obt_as.nchannels = obt.nchannels;
    obt_as.fmt = obt.fmt;
    obt_as.endianness = obt.endianness;

    audio_pcm_init_info (&hw->info, &obt_as);
    hw->samples = obt.samples;

    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
    if (!alsa->pcm_buf) {
        dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
               hw->samples, 1 << hw->info.shift);
        alsa_anal_close (&handle);
        return -1;
    }

    alsa->handle = handle;
    return 0;
}

static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
{
    int err;

    if (pause) {
        err = snd_pcm_drop (handle);
        if (err < 0) {
            alsa_logerr (err, "Could not stop %s\n", typ);
            return -1;
        }
    }
    else {
        err = snd_pcm_prepare (handle);
        if (err < 0) {
            alsa_logerr (err, "Could not prepare handle for %s\n", typ);
            return -1;
        }
    }

    return 0;
}

static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    switch (cmd) {
    case VOICE_ENABLE:
        ldebug ("enabling voice\n");
        return alsa_voice_ctl (alsa->handle, "playback", 0);

    case VOICE_DISABLE:
        ldebug ("disabling voice\n");
        return alsa_voice_ctl (alsa->handle, "playback", 1);
    }

    return -1;
}

static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
    struct alsa_params_req req;
    struct alsa_params_obt obt;
    snd_pcm_t *handle;
    struct audsettings obt_as;

    req.fmt = aud_to_alsafmt (as->fmt);
    req.freq = as->freq;
    req.nchannels = as->nchannels;
    req.period_size = conf.period_size_in;
    req.buffer_size = conf.buffer_size_in;
    req.size_in_usec = conf.size_in_usec_in;
    req.override_mask =
        (conf.period_size_in_overridden ? 1 : 0) |
        (conf.buffer_size_in_overridden ? 2 : 0);

    if (alsa_open (1, &req, &obt, &handle)) {
        return -1;
    }

    obt_as.freq = obt.freq;
    obt_as.nchannels = obt.nchannels;
    obt_as.fmt = obt.fmt;
    obt_as.endianness = obt.endianness;

    audio_pcm_init_info (&hw->info, &obt_as);
    hw->samples = obt.samples;

    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
    if (!alsa->pcm_buf) {
        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
               hw->samples, 1 << hw->info.shift);
        alsa_anal_close (&handle);
        return -1;
    }

    alsa->handle = handle;
    return 0;
}

static void alsa_fini_in (HWVoiceIn *hw)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;

    alsa_anal_close (&alsa->handle);

    if (alsa->pcm_buf) {
        qemu_free (alsa->pcm_buf);
        alsa->pcm_buf = NULL;
    }
}

static int alsa_run_in (HWVoiceIn *hw)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
    int hwshift = hw->info.shift;
    int i;
    int live = audio_pcm_hw_get_live_in (hw);
    int dead = hw->samples - live;
    int decr;
    struct {
        int add;
        int len;
    } bufs[2] = {
        { hw->wpos, 0 },
        { 0, 0 }
    };
    snd_pcm_sframes_t avail;
    snd_pcm_uframes_t read_samples = 0;

    if (!dead) {
        return 0;
    }

    avail = alsa_get_avail (alsa->handle);
    if (avail < 0) {
        dolog ("Could not get number of captured frames\n");
        return 0;
    }

    if (!avail) {
        snd_pcm_state_t state;

        state = snd_pcm_state (alsa->handle);
        switch (state) {
        case SND_PCM_STATE_PREPARED:
            avail = hw->samples;
            break;
        case SND_PCM_STATE_SUSPENDED:
            /* stream is suspended and waiting for an application recovery */
            if (alsa_resume (alsa->handle)) {
                dolog ("Failed to resume suspended input stream\n");
                return 0;
            }
            if (conf.verbose) {
                dolog ("Resuming suspended input stream\n");
            }
            break;
        default:
            if (conf.verbose) {
                dolog ("No frames available and ALSA state is %d\n", state);
            }
            return 0;
        }
    }

    decr = audio_MIN (dead, avail);
    if (!decr) {
        return 0;
    }

    if (hw->wpos + decr > hw->samples) {
        bufs[0].len = (hw->samples - hw->wpos);
        bufs[1].len = (decr - (hw->samples - hw->wpos));
    }
    else {
        bufs[0].len = decr;
    }

    for (i = 0; i < 2; ++i) {
        void *src;
        struct st_sample *dst;
        snd_pcm_sframes_t nread;
        snd_pcm_uframes_t len;

        len = bufs[i].len;

        src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
        dst = hw->conv_buf + bufs[i].add;

        while (len) {
            nread = snd_pcm_readi (alsa->handle, src, len);

            if (nread <= 0) {
                switch (nread) {
                case 0:
                    if (conf.verbose) {
                        dolog ("Failed to read %ld frames (read zero)\n", len);
                    }
                    goto exit;

                case -EPIPE:
                    if (alsa_recover (alsa->handle)) {
                        alsa_logerr (nread, "Failed to read %ld frames\n", len);
                        goto exit;
                    }
                    if (conf.verbose) {
                        dolog ("Recovering from capture xrun\n");
                    }
                    continue;

                case -EAGAIN:
                    goto exit;

                default:
                    alsa_logerr (
                        nread,
                        "Failed to read %ld frames from %p\n",
                        len,
                        src
                        );
                    goto exit;
                }
            }

            hw->conv (dst, src, nread, &nominal_volume);

            src = advance (src, nread << hwshift);
            dst += nread;

            read_samples += nread;
            len -= nread;
        }
    }

 exit:
    hw->wpos = (hw->wpos + read_samples) % hw->samples;
    return read_samples;
}

static int alsa_read (SWVoiceIn *sw, void *buf, int size)
{
    return audio_pcm_sw_read (sw, buf, size);
}

static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;

    switch (cmd) {
    case VOICE_ENABLE:
        ldebug ("enabling voice\n");
        return alsa_voice_ctl (alsa->handle, "capture", 0);

    case VOICE_DISABLE:
        ldebug ("disabling voice\n");
        return alsa_voice_ctl (alsa->handle, "capture", 1);
    }

    return -1;
}

static void *alsa_audio_init (void)
{
    return &conf;
}

static void alsa_audio_fini (void *opaque)
{
    (void) opaque;
}

static struct audio_option alsa_options[] = {
    {.name        = "DAC_SIZE_IN_USEC",
     .tag         = AUD_OPT_BOOL,
     .valp        = &conf.size_in_usec_out,
     .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"},
    {.name        = "DAC_PERIOD_SIZE",
     .tag         = AUD_OPT_INT,
     .valp        = &conf.period_size_out,
     .descr       = "DAC period size (0 to go with system default)",
     .overriddenp = &conf.period_size_out_overridden},
    {.name        = "DAC_BUFFER_SIZE",
     .tag         = AUD_OPT_INT,
     .valp        = &conf.buffer_size_out,
     .descr       = "DAC buffer size (0 to go with system default)",
     .overriddenp = &conf.buffer_size_out_overridden},
    {.name        = "ADC_SIZE_IN_USEC",
     .tag         = AUD_OPT_BOOL,
     .valp        = &conf.size_in_usec_in,
     .descr       = "ADC period/buffer size in microseconds (otherwise in frames)"},
    {.name        = "ADC_PERIOD_SIZE",
     .tag         = AUD_OPT_INT,
     .valp        = &conf.period_size_in,
     .descr       = "ADC period size (0 to go with system default)",
     .overriddenp = &conf.period_size_in_overridden},
    {.name        = "ADC_BUFFER_SIZE",
     .tag         = AUD_OPT_INT,
     .valp        = &conf.buffer_size_in,
     .descr       = "ADC buffer size (0 to go with system default)",
     .overriddenp = &conf.buffer_size_in_overridden},
    {.name        = "THRESHOLD",
     .tag         = AUD_OPT_INT,
     .valp        = &conf.threshold,
     .descr       = "(undocumented)"},
    {.name        = "DAC_DEV",
     .tag         = AUD_OPT_STR,
     .valp        = &conf.pcm_name_out,
     .descr       = "DAC device name (for instance dmix)"},
    {.name        = "ADC_DEV",
     .tag         = AUD_OPT_STR,
     .valp        = &conf.pcm_name_in,
     .descr       = "ADC device name"},
    {.name        = "VERBOSE",
     .tag         = AUD_OPT_BOOL,
     .valp        = &conf.verbose,
     .descr       = "Behave in a more verbose way"},
    { /* End of list */ }
};

static struct audio_pcm_ops alsa_pcm_ops = {
    alsa_init_out,
    alsa_fini_out,
    alsa_run_out,
    alsa_write,
    alsa_ctl_out,

    alsa_init_in,
    alsa_fini_in,
    alsa_run_in,
    alsa_read,
    alsa_ctl_in
};

struct audio_driver alsa_audio_driver = {
    .name           = "alsa",
    .descr          = "ALSA http://www.alsa-project.org",
    .options        = alsa_options,
    .init           = alsa_audio_init,
    .fini           = alsa_audio_fini,
    .pcm_ops        = &alsa_pcm_ops,
    .can_be_default = 1,
    .max_voices_out = INT_MAX,
    .max_voices_in  = INT_MAX,
    .voice_size_out = sizeof (ALSAVoiceOut),
    .voice_size_in  = sizeof (ALSAVoiceIn)
};