--- /dev/null
+/*
+ * QEMU ALSA audio driver
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include <alsa/asoundlib.h>
+#include "qemu-common.h"
+#include "qemu/main-loop.h"
+#include "audio.h"
+#include "trace.h"
+
+#if QEMU_GNUC_PREREQ(4, 3)
+#pragma GCC diagnostic ignored "-Waddress"
+#endif
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAConf {
+ int size_in_usec_in;
+ int size_in_usec_out;
+ const char *pcm_name_in;
+ const char *pcm_name_out;
+ unsigned int buffer_size_in;
+ unsigned int period_size_in;
+ unsigned int buffer_size_out;
+ unsigned int period_size_out;
+ unsigned int threshold;
+
+ int buffer_size_in_overridden;
+ int period_size_in_overridden;
+
+ int buffer_size_out_overridden;
+ int period_size_out_overridden;
+} ALSAConf;
+
+struct pollhlp {
+ snd_pcm_t *handle;
+ struct pollfd *pfds;
+ ALSAConf *conf;
+ int count;
+ int mask;
+};
+
+typedef struct ALSAVoiceOut {
+ HWVoiceOut hw;
+ int wpos;
+ int pending;
+ void *pcm_buf;
+ snd_pcm_t *handle;
+ struct pollhlp pollhlp;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+ HWVoiceIn hw;
+ snd_pcm_t *handle;
+ void *pcm_buf;
+ struct pollhlp pollhlp;
+} ALSAVoiceIn;
+
+struct alsa_params_req {
+ int freq;
+ snd_pcm_format_t fmt;
+ int nchannels;
+ int size_in_usec;
+ int override_mask;
+ unsigned int buffer_size;
+ unsigned int period_size;
+};
+
+struct alsa_params_obt {
+ int freq;
+ audfmt_e fmt;
+ int endianness;
+ int nchannels;
+ snd_pcm_uframes_t samples;
+};
+
+static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
+ int err,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void alsa_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 ("alsa run (prepared)");
+ break;
+
+ case SND_PCM_STATE_RUNNING:
+ audio_run ("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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+{
+ switch (fmt) {
+ case AUD_FMT_S8:
+ return SND_PCM_FORMAT_S8;
+
+ case AUD_FMT_U8:
+ return SND_PCM_FORMAT_U8;
+
+ case AUD_FMT_S16:
+ if (endianness) {
+ return SND_PCM_FORMAT_S16_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_S16_LE;
+ }
+
+ case AUD_FMT_U16:
+ if (endianness) {
+ return SND_PCM_FORMAT_U16_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_U16_LE;
+ }
+
+ case AUD_FMT_S32:
+ if (endianness) {
+ return SND_PCM_FORMAT_S32_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_S32_LE;
+ }
+
+ case AUD_FMT_U32:
+ if (endianness) {
+ return SND_PCM_FORMAT_U32_BE;
+ }
+ else {
+ return SND_PCM_FORMAT_U32_LE;
+ }
+
+ default:
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+ abort ();
+#endif
+ return SND_PCM_FORMAT_U8;
+ }
+}
+
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+ int *endianness)
+{
+ switch (alsafmt) {
+ case SND_PCM_FORMAT_S8:
+ *endianness = 0;
+ *fmt = AUD_FMT_S8;
+ break;
+
+ case SND_PCM_FORMAT_U8:
+ *endianness = 0;
+ *fmt = AUD_FMT_U8;
+ break;
+
+ case SND_PCM_FORMAT_S16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case SND_PCM_FORMAT_S16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case SND_PCM_FORMAT_S32_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S32;
+ break;
+
+ case SND_PCM_FORMAT_U32_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U32;
+ break;
+
+ case SND_PCM_FORMAT_S32_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S32;
+ break;
+
+ case SND_PCM_FORMAT_U32_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U32;
+ break;
+
+ default:
+ dolog ("Unrecognized audio format %d\n", alsafmt);
+ return -1;
+ }
+
+ return 0;
+}
+
+static void alsa_dump_info (struct alsa_params_req *req,
+ struct alsa_params_obt *obt,
+ snd_pcm_format_t obtfmt)
+{
+ 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 size %d period size %d\n",
+ req->buffer_size, req->period_size);
+ dolog ("obtained: samples %ld\n", obt->samples);
+}
+
+static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
+{
+ int err;
+ snd_pcm_sw_params_t *sw_params;
+
+ snd_pcm_sw_params_alloca (&sw_params);
+
+ err = snd_pcm_sw_params_current (handle, sw_params);
+ if (err < 0) {
+ dolog ("Could not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to get current software parameters\n");
+ return;
+ }
+
+ err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
+ if (err < 0) {
+ dolog ("Could not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software threshold to %ld\n",
+ threshold);
+ return;
+ }
+
+ err = snd_pcm_sw_params (handle, sw_params);
+ if (err < 0) {
+ dolog ("Could not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software parameters\n");
+ return;
+ }
+}
+
+static int alsa_open (int in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ ALSAConf *conf)
+{
+ snd_pcm_t *handle;
+ snd_pcm_hw_params_t *hw_params;
+ int err;
+ int size_in_usec;
+ unsigned int freq, nchannels;
+ const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+ snd_pcm_uframes_t obt_buffer_size;
+ const char *typ = in ? "ADC" : "DAC";
+ snd_pcm_format_t obtfmt;
+
+ freq = req->freq;
+ nchannels = req->nchannels;
+ size_in_usec = req->size_in_usec;
+
+ snd_pcm_hw_params_alloca (&hw_params);
+
+ err = snd_pcm_open (
+ &handle,
+ pcm_name,
+ in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
+ return -1;
+ }
+
+ err = snd_pcm_hw_params_any (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_access (
+ handle,
+ hw_params,
+ SND_PCM_ACCESS_RW_INTERLEAVED
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set access type\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
+ }
+
+ err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_channels_near (
+ handle,
+ hw_params,
+ &nchannels
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
+ req->nchannels);
+ goto err;
+ }
+
+ if (nchannels != 1 && nchannels != 2) {
+ alsa_logerr2 (err, typ,
+ "Can not handle obtained number of channels %d\n",
+ nchannels);
+ goto err;
+ }
+
+ if (req->buffer_size) {
+ unsigned long obt;
+
+ if (size_in_usec) {
+ int dir = 0;
+ unsigned int btime = req->buffer_size;
+
+ err = snd_pcm_hw_params_set_buffer_time_near (
+ handle,
+ hw_params,
+ &btime,
+ &dir
+ );
+ obt = btime;
+ }
+ else {
+ snd_pcm_uframes_t bsize = req->buffer_size;
+
+ err = snd_pcm_hw_params_set_buffer_size_near (
+ handle,
+ hw_params,
+ &bsize
+ );
+ obt = bsize;
+ }
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
+ size_in_usec ? "time" : "size", req->buffer_size);
+ goto err;
+ }
+
+ if ((req->override_mask & 2) && (obt - req->buffer_size))
+ dolog ("Requested buffer %s %u was rejected, using %lu\n",
+ size_in_usec ? "time" : "size", req->buffer_size, obt);
+ }
+
+ if (req->period_size) {
+ unsigned long obt;
+
+ if (size_in_usec) {
+ int dir = 0;
+ unsigned int ptime = req->period_size;
+
+ err = snd_pcm_hw_params_set_period_time_near (
+ handle,
+ hw_params,
+ &ptime,
+ &dir
+ );
+ obt = ptime;
+ }
+ else {
+ int dir = 0;
+ snd_pcm_uframes_t psize = req->period_size;
+
+ err = snd_pcm_hw_params_set_period_size_near (
+ handle,
+ hw_params,
+ &psize,
+ &dir
+ );
+ obt = psize;
+ }
+
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
+ size_in_usec ? "time" : "size", req->period_size);
+ goto err;
+ }
+
+ if (((req->override_mask & 1) && (obt - req->period_size)))
+ dolog ("Requested period %s %u was rejected, using %lu\n",
+ size_in_usec ? "time" : "size", req->period_size, obt);
+ }
+
+ err = snd_pcm_hw_params (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to get buffer size\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to get format\n");
+ goto err;
+ }
+
+ if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
+ dolog ("Invalid format was returned %d\n", obtfmt);
+ goto err;
+ }
+
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
+ goto err;
+ }
+
+ if (!in && conf->threshold) {
+ snd_pcm_uframes_t threshold;
+ int bytes_per_sec;
+
+ bytes_per_sec = freq << (nchannels == 2);
+
+ switch (obt->fmt) {
+ case AUD_FMT_S8:
+ case AUD_FMT_U8:
+ break;
+
+ case AUD_FMT_S16:
+ case AUD_FMT_U16:
+ bytes_per_sec <<= 1;
+ break;
+
+ case AUD_FMT_S32:
+ case AUD_FMT_U32:
+ bytes_per_sec <<= 2;
+ break;
+ }
+
+ threshold = (conf->threshold * bytes_per_sec) / 1000;
+ alsa_set_threshold (handle, threshold);
+ }
+
+ obt->nchannels = nchannels;
+ obt->freq = freq;
+ obt->samples = obt_buffer_size;
+
+ *handlep = handle;
+
+ if (obtfmt != req->fmt ||
+ obt->nchannels != req->nchannels ||
+ obt->freq != req->freq) {
+ dolog ("Audio parameters for %s\n", typ);
+ alsa_dump_info (req, obt, obtfmt);
+ }
+
+#ifdef DEBUG
+ alsa_dump_info (req, obt, obtfmt);
+#endif
+ return 0;
+
+ err:
+ alsa_anal_close1 (&handle);
+ return -1;
+}
+
+static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
+{
+ snd_pcm_sframes_t avail;
+
+ avail = snd_pcm_avail_update (handle);
+ if (avail < 0) {
+ if (avail == -EPIPE) {
+ if (!alsa_recover (handle)) {
+ avail = snd_pcm_avail_update (handle);
+ }
+ }
+
+ if (avail < 0) {
+ alsa_logerr (avail,
+ "Could not obtain number of available frames\n");
+ return -1;
+ }
+ }
+
+ return avail;
+}
+
+static void alsa_write_pending (ALSAVoiceOut *alsa)
+{
+ HWVoiceOut *hw = &alsa->hw;
+
+ while (alsa->pending) {
+ int left_till_end_samples = hw->samples - alsa->wpos;
+ int len = audio_MIN (alsa->pending, left_till_end_samples);
+ char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
+
+ while (len) {
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei (alsa->handle, src, len);
+
+ if (written <= 0) {
+ switch (written) {
+ case 0:
+ trace_alsa_wrote_zero(len);
+ return;
+
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (written, "Failed to write %d frames\n",
+ len);
+ return;
+ }
+ 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 %d frames\n",
+ len);
+ return;
+ }
+ trace_alsa_resume_out();
+ continue;
+
+ case -EAGAIN:
+ return;
+
+ default:
+ alsa_logerr (written, "Failed to write %d frames from %p\n",
+ len, src);
+ return;
+ }
+ }
+
+ alsa->wpos = (alsa->wpos + written) % hw->samples;
+ alsa->pending -= written;
+ len -= written;
+ }
+ }
+}
+
+static int alsa_run_out (HWVoiceOut *hw, int live)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ int decr;
+ snd_pcm_sframes_t avail;
+
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of available playback frames\n");
+ return 0;
+ }
+
+ decr = audio_MIN (live, avail);
+ decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
+ alsa->pending += decr;
+ alsa_write_pending (alsa);
+ return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ ldebug ("alsa_fini\n");
+ alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+
+ g_free(alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+}
+
+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;
+ ALSAConf *conf = drv_opaque;
+
+ req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
+ req.period_size = conf->period_size_out;
+ req.buffer_size = conf->buffer_size_out;
+ req.size_in_usec = conf->size_in_usec_out;
+ req.override_mask =
+ (conf->period_size_out_overridden ? 1 : 0) |
+ (conf->buffer_size_out_overridden ? 2 : 0);
+
+ if (alsa_open (0, &req, &obt, &handle, conf)) {
+ return -1;
+ }
+
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
+
+ audio_pcm_init_info (&hw->info, &obt_as);
+ hw->samples = obt.samples;
+
+ alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
+ if (!alsa->pcm_buf) {
+ dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ alsa_anal_close1 (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
+ 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 int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ {
+ va_list ap;
+ int poll_mode;
+
+ va_start (ap, cmd);
+ poll_mode = va_arg (ap, int);
+ va_end (ap);
+
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_out (hw)) {
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
+ return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
+ }
+
+ case VOICE_DISABLE:
+ ldebug ("disabling voice\n");
+ if (hw->poll_mode) {
+ hw->poll_mode = 0;
+ alsa_fini_poll (&alsa->pollhlp);
+ }
+ return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
+ }
+
+ return -1;
+}
+
+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;
+ ALSAConf *conf = drv_opaque;
+
+ req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
+ req.period_size = conf->period_size_in;
+ req.buffer_size = conf->buffer_size_in;
+ req.size_in_usec = conf->size_in_usec_in;
+ req.override_mask =
+ (conf->period_size_in_overridden ? 1 : 0) |
+ (conf->buffer_size_in_overridden ? 2 : 0);
+
+ if (alsa_open (1, &req, &obt, &handle, conf)) {
+ return -1;
+ }
+
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = obt.fmt;
+ obt_as.endianness = obt.endianness;
+
+ audio_pcm_init_info (&hw->info, &obt_as);
+ hw->samples = obt.samples;
+
+ alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+ if (!alsa->pcm_buf) {
+ dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ alsa_anal_close1 (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
+ return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ alsa_anal_close (&alsa->handle, &alsa->pollhlp);
+
+ g_free(alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+}
+
+static int alsa_run_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ int hwshift = hw->info.shift;
+ int i;
+ int live = audio_pcm_hw_get_live_in (hw);
+ int dead = hw->samples - live;
+ int decr;
+ struct {
+ int add;
+ int len;
+ } bufs[2] = {
+ { .add = hw->wpos, .len = 0 },
+ { .add = 0, .len = 0 }
+ };
+ snd_pcm_sframes_t avail;
+ snd_pcm_uframes_t read_samples = 0;
+
+ if (!dead) {
+ return 0;
+ }
+
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of captured frames\n");
+ return 0;
+ }
+
+ if (!avail) {
+ snd_pcm_state_t state;
+
+ state = snd_pcm_state (alsa->handle);
+ switch (state) {
+ case SND_PCM_STATE_PREPARED:
+ avail = hw->samples;
+ break;
+ case SND_PCM_STATE_SUSPENDED:
+ /* stream is suspended and waiting for an application recovery */
+ if (alsa_resume (alsa->handle)) {
+ dolog ("Failed to resume suspended input stream\n");
+ return 0;
+ }
+ trace_alsa_resume_in();
+ break;
+ default:
+ trace_alsa_no_frames(state);
+ return 0;
+ }
+ }
+
+ decr = audio_MIN (dead, avail);
+ if (!decr) {
+ return 0;
+ }
+
+ if (hw->wpos + decr > hw->samples) {
+ bufs[0].len = (hw->samples - hw->wpos);
+ bufs[1].len = (decr - (hw->samples - hw->wpos));
+ }
+ else {
+ bufs[0].len = decr;
+ }
+
+ for (i = 0; i < 2; ++i) {
+ void *src;
+ struct st_sample *dst;
+ snd_pcm_sframes_t nread;
+ snd_pcm_uframes_t len;
+
+ len = bufs[i].len;
+
+ src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
+ dst = hw->conv_buf + bufs[i].add;
+
+ while (len) {
+ nread = snd_pcm_readi (alsa->handle, src, len);
+
+ if (nread <= 0) {
+ switch (nread) {
+ case 0:
+ trace_alsa_read_zero(len);
+ goto exit;
+
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (nread, "Failed to read %ld frames\n", len);
+ goto exit;
+ }
+ trace_alsa_xrun_in();
+ continue;
+
+ case -EAGAIN:
+ goto exit;
+
+ default:
+ alsa_logerr (
+ nread,
+ "Failed to read %ld frames from %p\n",
+ len,
+ src
+ );
+ goto exit;
+ }
+ }
+
+ hw->conv (dst, src, nread);
+
+ src = advance (src, nread << hwshift);
+ dst += nread;
+
+ read_samples += nread;
+ len -= nread;
+ }
+ }
+
+ exit:
+ hw->wpos = (hw->wpos + read_samples) % hw->samples;
+ return read_samples;
+}
+
+static int alsa_read (SWVoiceIn *sw, void *buf, int size)
+{
+ return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ {
+ va_list ap;
+ int poll_mode;
+
+ va_start (ap, cmd);
+ poll_mode = va_arg (ap, int);
+ va_end (ap);
+
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_in (hw)) {
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
+
+ return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+ }
+
+ case VOICE_DISABLE:
+ ldebug ("disabling voice\n");
+ if (hw->poll_mode) {
+ hw->poll_mode = 0;
+ alsa_fini_poll (&alsa->pollhlp);
+ }
+ return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
+ }
+
+ return -1;
+}
+
+static ALSAConf glob_conf = {
+ .buffer_size_out = 4096,
+ .period_size_out = 1024,
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
+};
+
+static void *alsa_audio_init (void)
+{
+ ALSAConf *conf = g_malloc(sizeof(ALSAConf));
+ *conf = glob_conf;
+ return conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+ g_free(opaque);
+}
+
+static struct audio_option alsa_options[] = {
+ {
+ .name = "DAC_SIZE_IN_USEC",
+ .tag = AUD_OPT_BOOL,
+ .valp = &glob_conf.size_in_usec_out,
+ .descr = "DAC period/buffer size in microseconds (otherwise in frames)"
+ },
+ {
+ .name = "DAC_PERIOD_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.period_size_out,
+ .descr = "DAC period size (0 to go with system default)",
+ .overriddenp = &glob_conf.period_size_out_overridden
+ },
+ {
+ .name = "DAC_BUFFER_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_out,
+ .descr = "DAC buffer size (0 to go with system default)",
+ .overriddenp = &glob_conf.buffer_size_out_overridden
+ },
+ {
+ .name = "ADC_SIZE_IN_USEC",
+ .tag = AUD_OPT_BOOL,
+ .valp = &glob_conf.size_in_usec_in,
+ .descr =
+ "ADC period/buffer size in microseconds (otherwise in frames)"
+ },
+ {
+ .name = "ADC_PERIOD_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.period_size_in,
+ .descr = "ADC period size (0 to go with system default)",
+ .overriddenp = &glob_conf.period_size_in_overridden
+ },
+ {
+ .name = "ADC_BUFFER_SIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_in,
+ .descr = "ADC buffer size (0 to go with system default)",
+ .overriddenp = &glob_conf.buffer_size_in_overridden
+ },
+ {
+ .name = "THRESHOLD",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.threshold,
+ .descr = "(undocumented)"
+ },
+ {
+ .name = "DAC_DEV",
+ .tag = AUD_OPT_STR,
+ .valp = &glob_conf.pcm_name_out,
+ .descr = "DAC device name (for instance dmix)"
+ },
+ {
+ .name = "ADC_DEV",
+ .tag = AUD_OPT_STR,
+ .valp = &glob_conf.pcm_name_in,
+ .descr = "ADC device name"
+ },
+ { /* End of list */ }
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+ .init_out = alsa_init_out,
+ .fini_out = alsa_fini_out,
+ .run_out = alsa_run_out,
+ .write = alsa_write,
+ .ctl_out = alsa_ctl_out,
+
+ .init_in = alsa_init_in,
+ .fini_in = alsa_fini_in,
+ .run_in = alsa_run_in,
+ .read = alsa_read,
+ .ctl_in = alsa_ctl_in,
+};
+
+struct audio_driver alsa_audio_driver = {
+ .name = "alsa",
+ .descr = "ALSA http://www.alsa-project.org",
+ .options = alsa_options,
+ .init = alsa_audio_init,
+ .fini = alsa_audio_fini,
+ .pcm_ops = &alsa_pcm_ops,
+ .can_be_default = 1,
+ .max_voices_out = INT_MAX,
+ .max_voices_in = INT_MAX,
+ .voice_size_out = sizeof (ALSAVoiceOut),
+ .voice_size_in = sizeof (ALSAVoiceIn)
+};
Code Review / kvmfornfv.git / blobdiff