audio/alsaaudio.c - qemu - Git at Google

 /*
  * 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"

 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 (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);
     }

 #ifdef DEBUG
     alsa_dump_info(req, obt, obtfmt, pdo);
 #endif
     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,
     .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);
	/*
	* 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"

	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 (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);
	}

	#ifdef DEBUG
	alsa_dump_info(req, obt, obtfmt, pdo);
	#endif
	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,
	.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);