symbian-qemu-0.9.1-12/qemu-symbian-svp/audio/alsaaudio.c
changeset 1 2fb8b9db1c86
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/symbian-qemu-0.9.1-12/qemu-symbian-svp/audio/alsaaudio.c	Fri Jul 31 15:01:17 2009 +0100
@@ -0,0 +1,946 @@
+/*
+ * 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"
+
+#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 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 -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
+        | (!!conf.buffer_size_out_overridden << 1);
+
+    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
+        | (!!conf.buffer_size_in_overridden << 1);
+
+    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 (alsa->handle) == SND_PCM_STATE_PREPARED)) {
+        avail = hw->samples;
+    }
+
+    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[] = {
+    {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
+     "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
+     "DAC period size (0 to go with system default)",
+     &conf.period_size_out_overridden, 0},
+    {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+     "DAC buffer size (0 to go with system default)",
+     &conf.buffer_size_out_overridden, 0},
+
+    {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
+     "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+    {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
+     "ADC period size (0 to go with system default)",
+     &conf.period_size_in_overridden, 0},
+    {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+     "ADC buffer size (0 to go with system default)",
+     &conf.buffer_size_in_overridden, 0},
+
+    {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+     "(undocumented)", NULL, 0},
+
+    {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
+     "DAC device name (for instance dmix)", NULL, 0},
+
+    {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
+     "ADC device name", NULL, 0},
+
+    {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
+     "Behave in a more verbose way", NULL, 0},
+
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+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 = {
+    INIT_FIELD (name           = ) "alsa",
+    INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
+    INIT_FIELD (options        = ) alsa_options,
+    INIT_FIELD (init           = ) alsa_audio_init,
+    INIT_FIELD (fini           = ) alsa_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
+    INIT_FIELD (can_be_default = ) 1,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
+};