merged 15a_aqemu.patch audio patch (malc)

git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1584 c046a42c-6fe2-441c-8c8c-71466251a162

1 files changed, 926 insertions, 0 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
new file mode 100644
index 0000000000..133690576e
--- /dev/null
+++ b/audio/alsaaudio.c
@@ -0,0 +1,926 @@
+/*
+ * 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 "vl.h"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAVoiceOut {
+    HWVoiceOut hw;
+    void *pcm_buf;
+    snd_pcm_t *handle;
+    int can_pause;
+    int was_enabled;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+    HWVoiceIn hw;
+    snd_pcm_t *handle;
+    void *pcm_buf;
+    int can_pause;
+} 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_overriden;
+    int period_size_in_overriden;
+
+    int buffer_size_out_overriden;
+    int period_size_out_overriden;
+} conf = {
+#ifdef HIGH_LATENCY
+    .size_in_usec_in = 1,
+    .size_in_usec_out = 1,
+#endif
+    .pcm_name_out = "hw:0,0",
+    .pcm_name_in = "hw:0,0",
+#ifdef HIGH_LATENCY
+    .buffer_size_in = 400000,
+    .period_size_in = 400000 / 4,
+    .buffer_size_out = 400000,
+    .period_size_out = 400000 / 4,
+#else
+#define DEFAULT_BUFFER_SIZE 1024
+#define DEFAULT_PERIOD_SIZE 256
+    .buffer_size_in = DEFAULT_BUFFER_SIZE,
+    .period_size_in = DEFAULT_PERIOD_SIZE,
+    .buffer_size_out = DEFAULT_BUFFER_SIZE,
+    .period_size_out = DEFAULT_PERIOD_SIZE,
+    .buffer_size_in_overriden = 0,
+    .buffer_size_out_overriden = 0,
+    .period_size_in_overriden = 0,
+    .period_size_out_overriden = 0,
+#endif
+    .threshold = 0
+};
+
+struct alsa_params_req {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    unsigned int buffer_size;
+    unsigned int period_size;
+};
+
+struct alsa_params_obt {
+    int freq;
+    audfmt_e fmt;
+    int nchannels;
+    int can_pause;
+    snd_pcm_uframes_t buffer_size;
+};
+
+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, "Can 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 int 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;
+
+    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 (int 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;
+
+    default:
+        dolog ("Unrecognized audio format %d\n", alsafmt);
+        return -1;
+    }
+
+    return 0;
+}
+
+#ifdef DEBUG_MISMATCHES
+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: buffer size %ld\n", obt->buffer_size);
+}
+#endif
+
+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 ("Can 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 ("Can 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 ("Can 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, freq, nchannels;
+    const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+    unsigned int period_size, buffer_size;
+    snd_pcm_uframes_t obt_buffer_size;
+    const char *typ = in ? "ADC" : "DAC";
+
+    freq = req->freq;
+    period_size = req->period_size;
+    buffer_size = req->buffer_size;
+    nchannels = req->nchannels;
+
+    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) {
+        alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
+        goto err;
+    }
+
+    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 (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
+        if (!buffer_size) {
+            buffer_size = DEFAULT_BUFFER_SIZE;
+            period_size= DEFAULT_PERIOD_SIZE;
+        }
+    }
+
+    if (buffer_size) {
+        if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
+            if (period_size) {
+                err = snd_pcm_hw_params_set_period_time_near (
+                    handle,
+                    hw_params,
+                    &period_size,
+                    0);
+                if (err < 0) {
+                    alsa_logerr2 (err, typ,
+                                  "Failed to set period time %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_time_near (
+                handle,
+                hw_params,
+                &buffer_size,
+                0);
+
+            if (err < 0) {
+                alsa_logerr2 (err, typ,
+                              "Failed to set buffer time %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+        else {
+            int dir;
+            snd_pcm_uframes_t minval;
+
+            if (period_size) {
+                minval = period_size;
+                dir = 0;
+
+                err = snd_pcm_hw_params_get_period_size_min (
+                    hw_params,
+                    &minval,
+                    &dir
+                    );
+                if (err < 0) {
+                    alsa_logerr (
+                        err,
+                        "Can not get minmal period size for %s\n",
+                        typ
+                        );
+                }
+                else {
+                    if (period_size < minval) {
+                        if ((in && conf.period_size_in_overriden)
+                            || (!in && conf.period_size_out_overriden)) {
+                            dolog ("%s period size(%d) is less "
+                                   "than minmal period size(%ld)\n",
+                                   typ,
+                                   period_size,
+                                   minval);
+                        }
+                        period_size = minval;
+                    }
+                }
+
+                err = snd_pcm_hw_params_set_period_size (
+                    handle,
+                    hw_params,
+                    period_size,
+                    0
+                    );
+                if (err < 0) {
+                    alsa_logerr2 (err, typ, "Failed to set period size %d\n",
+                                  req->period_size);
+                    goto err;
+                }
+            }
+
+            minval = buffer_size;
+            err = snd_pcm_hw_params_get_buffer_size_min (
+                hw_params,
+                &minval
+                );
+            if (err < 0) {
+                alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+                             typ);
+            }
+            else {
+                if (buffer_size < minval) {
+                    if ((in && conf.buffer_size_in_overriden)
+                        || (!in && conf.buffer_size_out_overriden)) {
+                        dolog (
+                            "%s buffer size(%d) is less "
+                            "than minimal buffer size(%ld)\n",
+                            typ,
+                            buffer_size,
+                            minval
+                            );
+                    }
+                    buffer_size = minval;
+                }
+            }
+
+            err = snd_pcm_hw_params_set_buffer_size (
+                handle,
+                hw_params,
+                buffer_size
+                );
+            if (err < 0) {
+                alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
+                              req->buffer_size);
+                goto err;
+            }
+        }
+    }
+    else {
+        dolog ("warning: buffer size is not set\n");
+    }
+
+    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_prepare (handle);
+    if (err < 0) {
+        alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+        goto err;
+    }
+
+    obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
+    if (obt->can_pause < 0) {
+        alsa_logerr (err, "Can not get pause capability for %s\n", typ);
+        obt->can_pause = 0;
+    }
+
+    if (!in && conf.threshold) {
+        snd_pcm_uframes_t threshold;
+        int bytes_per_sec;
+
+        bytes_per_sec = freq
+            << (nchannels == 2)
+            << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+
+        threshold = (conf.threshold * bytes_per_sec) / 1000;
+        alsa_set_threshold (handle, threshold);
+    }
+
+    obt->fmt = req->fmt;
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+    *handlep = handle;
+
+    if (obt->fmt != req->fmt ||
+        obt->nchannels != req->nchannels ||
+        obt->freq != req->freq) {
+#ifdef DEBUG_MISMATCHES
+        dolog ("Audio paramters mismatch for %s\n", typ);
+        alsa_dump_info (req, obt);
+#endif
+    }
+
+#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_run_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    int rpos, live, decr;
+    int samples;
+    uint8_t *dst;
+    st_sample_t *src;
+    snd_pcm_sframes_t avail;
+
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
+
+    avail = snd_pcm_avail_update (alsa->handle);
+    if (avail < 0) {
+        if (avail == -EPIPE) {
+            if (!alsa_recover (alsa->handle)) {
+                avail = snd_pcm_avail_update (alsa->handle);
+                if (avail >= 0) {
+                    goto ok;
+                }
+            }
+        }
+
+        alsa_logerr (avail, "Can not get amount free space\n");
+        return 0;
+    }
+
+ ok:
+    decr = audio_MIN (live, avail);
+    samples = decr;
+    rpos = hw->rpos;
+    while (samples) {
+        int left_till_end_samples = hw->samples - rpos;
+        int convert_samples = 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, convert_samples);
+
+    again:
+        written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+
+        if (written < 0) {
+            switch (written) {
+            case -EPIPE:
+                if (!alsa_recover (alsa->handle)) {
+                    goto again;
+                }
+                dolog (
+                    "Failed to write %d frames to %p, handle %p not prepared\n",
+                    convert_samples,
+                    dst,
+                    alsa->handle
+                    );
+                goto exit;
+
+            case -EAGAIN:
+                goto again;
+
+            default:
+                alsa_logerr (written, "Failed to write %d frames to %p\n",
+                             convert_samples, dst);
+                goto exit;
+            }
+        }
+
+        mixeng_clear (src, written);
+        rpos = (rpos + written) % hw->samples;
+        samples -= 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, int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    audfmt_e effective_fmt;
+    int endianness;
+    int err;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_out;
+    req.buffer_size = conf.buffer_size_out;
+
+    if (alsa_open (0, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    alsa->handle = handle;
+    alsa->was_enabled = 0;
+    return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    int err;
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    switch (cmd) {
+    case VOICE_ENABLE:
+        ldebug ("enabling voice\n");
+        audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
+        if (alsa->can_pause) {
+            /* Why this was_enabled madness is needed at all?? */
+            if (alsa->was_enabled) {
+                err = snd_pcm_pause (alsa->handle, 0);
+                if (err < 0) {
+                    alsa_logerr (err, "Failed to resume playing\n");
+                    /* not fatal really */
+                }
+            }
+            else {
+                alsa->was_enabled = 1;
+            }
+        }
+        break;
+
+    case VOICE_DISABLE:
+        ldebug ("disabling voice\n");
+        if (alsa->can_pause) {
+            err = snd_pcm_pause (alsa->handle, 1);
+            if (err < 0) {
+                alsa_logerr (err, "Failed to stop playing\n");
+                /* not fatal really */
+            }
+        }
+        break;
+    }
+    return 0;
+}
+
+static int alsa_init_in (HWVoiceIn *hw,
+                        int freq, int nchannels, audfmt_e fmt)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    int endianness;
+    int err;
+    audfmt_e effective_fmt;
+    snd_pcm_t *handle;
+
+    req.fmt = aud_to_alsafmt (fmt);
+    req.freq = freq;
+    req.nchannels = nchannels;
+    req.period_size = conf.period_size_in;
+    req.buffer_size = conf.buffer_size_in;
+
+    if (alsa_open (1, &req, &obt, &handle)) {
+        return -1;
+    }
+
+    err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+    if (err) {
+        alsa_anal_close (&handle);
+        return -1;
+    }
+
+    audio_pcm_init_info (
+        &hw->info,
+        obt.freq,
+        obt.nchannels,
+        effective_fmt,
+        audio_need_to_swap_endian (endianness)
+        );
+    alsa->can_pause = obt.can_pause;
+    hw->bufsize = obt.buffer_size;
+    alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+    if (!alsa->pcm_buf) {
+        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;
+    struct {
+        int add;
+        int len;
+    } bufs[2] = {
+        { hw->wpos, 0 },
+        { 0, 0 }
+    };
+
+    snd_pcm_uframes_t read_samples = 0;
+
+    if (!dead) {
+        return 0;
+    }
+
+    if (hw->wpos + dead > hw->samples) {
+        bufs[0].len = (hw->samples - hw->wpos);
+        bufs[1].len = (dead - (hw->samples - hw->wpos));
+    }
+    else {
+        bufs[0].len = dead;
+    }
+
+
+    for (i = 0; i < 2; ++i) {
+        void *src;
+        st_sample_t *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 -EPIPE:
+                    if (!alsa_recover (alsa->handle)) {
+                        continue;
+                    }
+                    dolog (
+                        "Failed to read %ld frames from %p, "
+                        "handle %p not prepared\n",
+                        len,
+                        src,
+                        alsa->handle
+                        );
+                    goto exit;
+
+                case -EAGAIN:
+                    continue;
+
+                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, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+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", &conf.period_size_out_overriden, 0},
+    {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+     "DAC buffer size", &conf.buffer_size_out_overriden, 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", &conf.period_size_in_overriden, 0},
+    {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+     "ADC buffer size", &conf.buffer_size_in_overriden, 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},
+    {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)
+};