Introduce Virtio-loopback epsilon release:

Epsilon release introduces a new compatibility layer which make virtio-loopback
design to work with QEMU and rust-vmm vhost-user backend without require any
changes.

Signed-off-by: Timos Ampelikiotis <t.ampelikiotis@virtualopensystems.com>
Change-Id: I52e57563e08a7d0bdc002f8e928ee61ba0c53dd9

1 files changed, 946 insertions, 0 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
new file mode 100644
index 000000000..2b9789e64
--- /dev/null
+++ b/audio/alsaaudio.c
@@ -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 "qemu/osdep.h"
+#include <alsa/asoundlib.h>
+#include "qemu/main-loop.h"
+#include "qemu/module.h"
+#include "audio.h"
+#include "trace.h"
+
+#pragma GCC diagnostic ignored "-Waddress"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+#define DEBUG_ALSA 0
+
+struct pollhlp {
+    snd_pcm_t *handle;
+    struct pollfd *pfds;
+    int count;
+    int mask;
+    AudioState *s;
+};
+
+typedef struct ALSAVoiceOut {
+    HWVoiceOut hw;
+    snd_pcm_t *handle;
+    struct pollhlp pollhlp;
+    Audiodev *dev;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+    HWVoiceIn hw;
+    snd_pcm_t *handle;
+    struct pollhlp pollhlp;
+    Audiodev *dev;
+} ALSAVoiceIn;
+
+struct alsa_params_req {
+    int freq;
+    snd_pcm_format_t fmt;
+    int nchannels;
+};
+
+struct alsa_params_obt {
+    int freq;
+    AudioFormat 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_fini_poll (struct pollhlp *hlp)
+{
+    int i;
+    struct pollfd *pfds = hlp->pfds;
+
+    if (pfds) {
+        for (i = 0; i < hlp->count; ++i) {
+            qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
+        }
+        g_free (pfds);
+    }
+    hlp->pfds = NULL;
+    hlp->count = 0;
+    hlp->handle = NULL;
+}
+
+static void alsa_anal_close1 (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 void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
+{
+    alsa_fini_poll (hlp);
+    alsa_anal_close1 (handlep);
+}
+
+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 void alsa_poll_handler (void *opaque)
+{
+    int err, count;
+    snd_pcm_state_t state;
+    struct pollhlp *hlp = opaque;
+    unsigned short revents;
+
+    count = poll (hlp->pfds, hlp->count, 0);
+    if (count < 0) {
+        dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
+        return;
+    }
+
+    if (!count) {
+        return;
+    }
+
+    /* XXX: ALSA example uses initial count, not the one returned by
+       poll, correct? */
+    err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
+                                            hlp->count, &revents);
+    if (err < 0) {
+        alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
+        return;
+    }
+
+    if (!(revents & hlp->mask)) {
+        trace_alsa_revents(revents);
+        return;
+    }
+
+    state = snd_pcm_state (hlp->handle);
+    switch (state) {
+    case SND_PCM_STATE_SETUP:
+        alsa_recover (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_XRUN:
+        alsa_recover (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_SUSPENDED:
+        alsa_resume (hlp->handle);
+        break;
+
+    case SND_PCM_STATE_PREPARED:
+        audio_run(hlp->s, "alsa run (prepared)");
+        break;
+
+    case SND_PCM_STATE_RUNNING:
+        audio_run(hlp->s, "alsa run (running)");
+        break;
+
+    default:
+        dolog ("Unexpected state %d\n", state);
+    }
+}
+
+static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
+{
+    int i, count, err;
+    struct pollfd *pfds;
+
+    count = snd_pcm_poll_descriptors_count (handle);
+    if (count <= 0) {
+        dolog ("Could not initialize poll mode\n"
+               "Invalid number of poll descriptors %d\n", count);
+        return -1;
+    }
+
+    pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
+    if (!pfds) {
+        dolog ("Could not initialize poll mode\n");
+        return -1;
+    }
+
+    err = snd_pcm_poll_descriptors (handle, pfds, count);
+    if (err < 0) {
+        alsa_logerr (err, "Could not initialize poll mode\n"
+                     "Could not obtain poll descriptors\n");
+        g_free (pfds);
+        return -1;
+    }
+
+    for (i = 0; i < count; ++i) {
+        if (pfds[i].events & POLLIN) {
+            qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
+        }
+        if (pfds[i].events & POLLOUT) {
+            trace_alsa_pollout(i, pfds[i].fd);
+            qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
+        }
+        trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
+
+    }
+    hlp->pfds = pfds;
+    hlp->count = count;
+    hlp->handle = handle;
+    hlp->mask = mask;
+    return 0;
+}
+
+static int alsa_poll_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
+}
+
+static int alsa_poll_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
+}
+
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
+{
+    switch (fmt) {
+    case AUDIO_FORMAT_S8:
+        return SND_PCM_FORMAT_S8;
+
+    case AUDIO_FORMAT_U8:
+        return SND_PCM_FORMAT_U8;
+
+    case AUDIO_FORMAT_S16:
+        if (endianness) {
+            return SND_PCM_FORMAT_S16_BE;
+        } else {
+            return SND_PCM_FORMAT_S16_LE;
+        }
+
+    case AUDIO_FORMAT_U16:
+        if (endianness) {
+            return SND_PCM_FORMAT_U16_BE;
+        } else {
+            return SND_PCM_FORMAT_U16_LE;
+        }
+
+    case AUDIO_FORMAT_S32:
+        if (endianness) {
+            return SND_PCM_FORMAT_S32_BE;
+        } else {
+            return SND_PCM_FORMAT_S32_LE;
+        }
+
+    case AUDIO_FORMAT_U32:
+        if (endianness) {
+            return SND_PCM_FORMAT_U32_BE;
+        } else {
+            return SND_PCM_FORMAT_U32_LE;
+        }
+
+    case AUDIO_FORMAT_F32:
+        if (endianness) {
+            return SND_PCM_FORMAT_FLOAT_BE;
+        } else {
+            return SND_PCM_FORMAT_FLOAT_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, AudioFormat *fmt,
+                           int *endianness)
+{
+    switch (alsafmt) {
+    case SND_PCM_FORMAT_S8:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_S8;
+        break;
+
+    case SND_PCM_FORMAT_U8:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_U8;
+        break;
+
+    case SND_PCM_FORMAT_S16_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_S16;
+        break;
+
+    case SND_PCM_FORMAT_U16_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_U16;
+        break;
+
+    case SND_PCM_FORMAT_S16_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_S16;
+        break;
+
+    case SND_PCM_FORMAT_U16_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_U16;
+        break;
+
+    case SND_PCM_FORMAT_S32_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_U32;
+        break;
+
+    case SND_PCM_FORMAT_S32_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_U32;
+        break;
+
+    case SND_PCM_FORMAT_FLOAT_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_F32;
+        break;
+
+    case SND_PCM_FORMAT_FLOAT_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_F32;
+        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,
+                            snd_pcm_format_t obtfmt,
+                            AudiodevAlsaPerDirectionOptions *apdo)
+{
+    dolog("parameter | requested value | obtained value\n");
+    dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
+    dolog("channels  |      %10d |     %10d\n",
+          req->nchannels, obt->nchannels);
+    dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
+    dolog("============================================\n");
+    dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
+          apdo->buffer_length, apdo->period_length);
+    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(bool in, struct alsa_params_req *req,
+                     struct alsa_params_obt *obt, snd_pcm_t **handlep,
+                     Audiodev *dev)
+{
+    AudiodevAlsaOptions *aopts = &dev->u.alsa;
+    AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
+    snd_pcm_t *handle;
+    snd_pcm_hw_params_t *hw_params;
+    int err;
+    unsigned int freq, nchannels;
+    const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
+    snd_pcm_uframes_t obt_buffer_size;
+    const char *typ = in ? "ADC" : "DAC";
+    snd_pcm_format_t obtfmt;
+
+    freq = req->freq;
+    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);
+    }
+
+    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 (apdo->buffer_length) {
+        int dir = 0;
+        unsigned int btime = apdo->buffer_length;
+
+        err = snd_pcm_hw_params_set_buffer_time_near(
+            handle, hw_params, &btime, &dir);
+
+        if (err < 0) {
+            alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
+                         apdo->buffer_length);
+            goto err;
+        }
+
+        if (apdo->has_buffer_length && btime != apdo->buffer_length) {
+            dolog("Requested buffer time %" PRId32
+                  " was rejected, using %u\n", apdo->buffer_length, btime);
+        }
+    }
+
+    if (apdo->period_length) {
+        int dir = 0;
+        unsigned int ptime = apdo->period_length;
+
+        err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+                                                     &dir);
+
+        if (err < 0) {
+            alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
+                         apdo->period_length);
+            goto err;
+        }
+
+        if (apdo->has_period_length && ptime != apdo->period_length) {
+            dolog("Requested period time %" PRId32 " was rejected, using %d\n",
+                  apdo->period_length, ptime);
+        }
+    }
+
+    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 && aopts->has_threshold && aopts->threshold) {
+        struct audsettings as = { .freq = freq };
+        alsa_set_threshold(
+            handle,
+            audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
+                                &as, aopts->threshold));
+    }
+
+    obt->nchannels = nchannels;
+    obt->freq = freq;
+    obt->samples = obt_buffer_size;
+
+    *handlep = handle;
+
+    if (DEBUG_ALSA || obtfmt != req->fmt ||
+        obt->nchannels != req->nchannels || obt->freq != req->freq) {
+        dolog ("Audio parameters for %s\n", typ);
+        alsa_dump_info(req, obt, obtfmt, apdo);
+    }
+
+    return 0;
+
+ err:
+    alsa_anal_close1 (&handle);
+    return -1;
+}
+
+static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    size_t pos = 0;
+    size_t len_frames = len / hw->info.bytes_per_frame;
+
+    while (len_frames) {
+        char *src = advance(buf, pos);
+        snd_pcm_sframes_t written;
+
+        written = snd_pcm_writei(alsa->handle, src, len_frames);
+
+        if (written <= 0) {
+            switch (written) {
+            case 0:
+                trace_alsa_wrote_zero(len_frames);
+                return pos;
+
+            case -EPIPE:
+                if (alsa_recover(alsa->handle)) {
+                    alsa_logerr(written, "Failed to write %zu frames\n",
+                                len_frames);
+                    return pos;
+                }
+                trace_alsa_xrun_out();
+                continue;
+
+            case -ESTRPIPE:
+                /*
+                 * stream is suspended and waiting for an application
+                 * recovery
+                 */
+                if (alsa_resume(alsa->handle)) {
+                    alsa_logerr(written, "Failed to write %zu frames\n",
+                                len_frames);
+                    return pos;
+                }
+                trace_alsa_resume_out();
+                continue;
+
+            case -EAGAIN:
+                return pos;
+
+            default:
+                alsa_logerr(written, "Failed to write %zu frames from %p\n",
+                            len, src);
+                return pos;
+            }
+        }
+
+        pos += written * hw->info.bytes_per_frame;
+        if (written < len_frames) {
+            break;
+        }
+        len_frames -= written;
+    }
+
+    return pos;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+    ldebug ("alsa_fini\n");
+    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+}
+
+static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
+                         void *drv_opaque)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    snd_pcm_t *handle;
+    struct audsettings obt_as;
+    Audiodev *dev = drv_opaque;
+
+    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+
+    if (alsa_open(0, &req, &obt, &handle, dev)) {
+        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->pollhlp.s = hw->s;
+    alsa->handle = handle;
+    alsa->dev = dev;
+    return 0;
+}
+
+#define VOICE_CTL_PAUSE 0
+#define VOICE_CTL_PREPARE 1
+#define VOICE_CTL_START 2
+
+static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
+{
+    int err;
+
+    if (ctl == VOICE_CTL_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;
+        }
+        if (ctl == VOICE_CTL_START) {
+            err = snd_pcm_start(handle);
+            if (err < 0) {
+                alsa_logerr (err, "Could not start handle for %s\n", typ);
+                return -1;
+            }
+        }
+    }
+
+    return 0;
+}
+
+static void alsa_enable_out(HWVoiceOut *hw, bool enable)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
+
+    if (enable) {
+        bool poll_mode = apdo->try_poll;
+
+        ldebug("enabling voice\n");
+        if (poll_mode && alsa_poll_out(hw)) {
+            poll_mode = 0;
+        }
+        hw->poll_mode = poll_mode;
+        alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
+    } else {
+        ldebug("disabling voice\n");
+        if (hw->poll_mode) {
+            hw->poll_mode = 0;
+            alsa_fini_poll(&alsa->pollhlp);
+        }
+        alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
+    }
+}
+
+static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    struct alsa_params_req req;
+    struct alsa_params_obt obt;
+    snd_pcm_t *handle;
+    struct audsettings obt_as;
+    Audiodev *dev = drv_opaque;
+
+    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+    req.freq = as->freq;
+    req.nchannels = as->nchannels;
+
+    if (alsa_open(1, &req, &obt, &handle, dev)) {
+        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->pollhlp.s = hw->s;
+    alsa->handle = handle;
+    alsa->dev = dev;
+    return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+    alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+}
+
+static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    size_t pos = 0;
+
+    while (len) {
+        void *dst = advance(buf, pos);
+        snd_pcm_sframes_t nread;
+
+        nread = snd_pcm_readi(
+            alsa->handle, dst, len / hw->info.bytes_per_frame);
+
+        if (nread <= 0) {
+            switch (nread) {
+            case 0:
+                trace_alsa_read_zero(len);
+                return pos;
+
+            case -EPIPE:
+                if (alsa_recover(alsa->handle)) {
+                    alsa_logerr(nread, "Failed to read %zu frames\n", len);
+                    return pos;
+                }
+                trace_alsa_xrun_in();
+                continue;
+
+            case -EAGAIN:
+                return pos;
+
+            default:
+                alsa_logerr(nread, "Failed to read %zu frames to %p\n",
+                            len, dst);
+                return pos;
+            }
+        }
+
+        pos += nread * hw->info.bytes_per_frame;
+        len -= nread * hw->info.bytes_per_frame;
+    }
+
+    return pos;
+}
+
+static void alsa_enable_in(HWVoiceIn *hw, bool enable)
+{
+    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
+
+    if (enable) {
+        bool poll_mode = apdo->try_poll;
+
+        ldebug("enabling voice\n");
+        if (poll_mode && alsa_poll_in(hw)) {
+            poll_mode = 0;
+        }
+        hw->poll_mode = poll_mode;
+
+        alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
+    } else {
+        ldebug ("disabling voice\n");
+        if (hw->poll_mode) {
+            hw->poll_mode = 0;
+            alsa_fini_poll(&alsa->pollhlp);
+        }
+        alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
+    }
+}
+
+static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
+{
+    if (!apdo->has_try_poll) {
+        apdo->try_poll = true;
+        apdo->has_try_poll = true;
+    }
+}
+
+static void *alsa_audio_init(Audiodev *dev)
+{
+    AudiodevAlsaOptions *aopts;
+    assert(dev->driver == AUDIODEV_DRIVER_ALSA);
+
+    aopts = &dev->u.alsa;
+    alsa_init_per_direction(aopts->in);
+    alsa_init_per_direction(aopts->out);
+
+    /*
+     * need to define them, as otherwise alsa produces no sound
+     * doesn't set has_* so alsa_open can identify it wasn't set by the user
+     */
+    if (!dev->u.alsa.out->has_period_length) {
+        /* 1024 frames assuming 44100Hz */
+        dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
+    }
+    if (!dev->u.alsa.out->has_buffer_length) {
+        /* 4096 frames assuming 44100Hz */
+        dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
+    }
+
+    /*
+     * OptsVisitor sets unspecified optional fields to zero, but do not depend
+     * on it...
+     */
+    if (!dev->u.alsa.in->has_period_length) {
+        dev->u.alsa.in->period_length = 0;
+    }
+    if (!dev->u.alsa.in->has_buffer_length) {
+        dev->u.alsa.in->buffer_length = 0;
+    }
+
+    return dev;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+}
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+    .init_out = alsa_init_out,
+    .fini_out = alsa_fini_out,
+    .write    = alsa_write,
+    .run_buffer_out = audio_generic_run_buffer_out,
+    .enable_out = alsa_enable_out,
+
+    .init_in  = alsa_init_in,
+    .fini_in  = alsa_fini_in,
+    .read     = alsa_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
+    .enable_in = alsa_enable_in,
+};
+
+static struct audio_driver alsa_audio_driver = {
+    .name           = "alsa",
+    .descr          = "ALSA http://www.alsa-project.org",
+    .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)
+};
+
+static void register_audio_alsa(void)
+{
+    audio_driver_register(&alsa_audio_driver);
+}
+type_init(register_audio_alsa);