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authorbellard <bellard@c046a42c-6fe2-441c-8c8c-71466251a162>2005-10-30 18:58:22 +0000
committerbellard <bellard@c046a42c-6fe2-441c-8c8c-71466251a162>2005-10-30 18:58:22 +0000
commit1d14ffa97eacd3cb722271eaf6f093038396eac4 (patch)
tree1aae1f090262c3642cc672971890141050413d26 /audio/alsaaudio.c
parent3b0d4f61c917c4612b561d75b33a11f4da00738b (diff)
merged 15a_aqemu.patch audio patch (malc)
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1584 c046a42c-6fe2-441c-8c8c-71466251a162
Diffstat (limited to 'audio/alsaaudio.c')
-rw-r--r--audio/alsaaudio.c926
1 files changed, 926 insertions, 0 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
new file mode 100644
index 0000000000..133690576e
--- /dev/null
+++ b/audio/alsaaudio.c
@@ -0,0 +1,926 @@
+/*
+ * QEMU ALSA audio driver
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include <alsa/asoundlib.h>
+#include "vl.h"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAVoiceOut {
+ HWVoiceOut hw;
+ void *pcm_buf;
+ snd_pcm_t *handle;
+ int can_pause;
+ int was_enabled;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+ HWVoiceIn hw;
+ snd_pcm_t *handle;
+ void *pcm_buf;
+ int can_pause;
+} ALSAVoiceIn;
+
+static struct {
+ int size_in_usec_in;
+ int size_in_usec_out;
+ const char *pcm_name_in;
+ const char *pcm_name_out;
+ unsigned int buffer_size_in;
+ unsigned int period_size_in;
+ unsigned int buffer_size_out;
+ unsigned int period_size_out;
+ unsigned int threshold;
+
+ int buffer_size_in_overriden;
+ int period_size_in_overriden;
+
+ int buffer_size_out_overriden;
+ int period_size_out_overriden;
+} conf = {
+#ifdef HIGH_LATENCY
+ .size_in_usec_in = 1,
+ .size_in_usec_out = 1,
+#endif
+ .pcm_name_out = "hw:0,0",
+ .pcm_name_in = "hw:0,0",
+#ifdef HIGH_LATENCY
+ .buffer_size_in = 400000,
+ .period_size_in = 400000 / 4,
+ .buffer_size_out = 400000,
+ .period_size_out = 400000 / 4,
+#else
+#define DEFAULT_BUFFER_SIZE 1024
+#define DEFAULT_PERIOD_SIZE 256
+ .buffer_size_in = DEFAULT_BUFFER_SIZE,
+ .period_size_in = DEFAULT_PERIOD_SIZE,
+ .buffer_size_out = DEFAULT_BUFFER_SIZE,
+ .period_size_out = DEFAULT_PERIOD_SIZE,
+ .buffer_size_in_overriden = 0,
+ .buffer_size_out_overriden = 0,
+ .period_size_in_overriden = 0,
+ .period_size_out_overriden = 0,
+#endif
+ .threshold = 0
+};
+
+struct alsa_params_req {
+ int freq;
+ audfmt_e fmt;
+ int nchannels;
+ unsigned int buffer_size;
+ unsigned int period_size;
+};
+
+struct alsa_params_obt {
+ int freq;
+ audfmt_e fmt;
+ int nchannels;
+ int can_pause;
+ snd_pcm_uframes_t buffer_size;
+};
+
+static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
+ int err,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void alsa_anal_close (snd_pcm_t **handlep)
+{
+ int err = snd_pcm_close (*handlep);
+ if (err) {
+ alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
+ }
+ *handlep = NULL;
+}
+
+static int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_alsafmt (audfmt_e fmt)
+{
+ switch (fmt) {
+ case AUD_FMT_S8:
+ return SND_PCM_FORMAT_S8;
+
+ case AUD_FMT_U8:
+ return SND_PCM_FORMAT_U8;
+
+ case AUD_FMT_S16:
+ return SND_PCM_FORMAT_S16_LE;
+
+ case AUD_FMT_U16:
+ return SND_PCM_FORMAT_U16_LE;
+
+ default:
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+ abort ();
+#endif
+ return SND_PCM_FORMAT_U8;
+ }
+}
+
+static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
+{
+ switch (alsafmt) {
+ case SND_PCM_FORMAT_S8:
+ *endianness = 0;
+ *fmt = AUD_FMT_S8;
+ break;
+
+ case SND_PCM_FORMAT_U8:
+ *endianness = 0;
+ *fmt = AUD_FMT_U8;
+ break;
+
+ case SND_PCM_FORMAT_S16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case SND_PCM_FORMAT_S16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ default:
+ dolog ("Unrecognized audio format %d\n", alsafmt);
+ return -1;
+ }
+
+ return 0;
+}
+
+#ifdef DEBUG_MISMATCHES
+static void alsa_dump_info (struct alsa_params_req *req,
+ struct alsa_params_obt *obt)
+{
+ dolog ("parameter | requested value | obtained value\n");
+ dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
+ dolog ("channels | %10d | %10d\n",
+ req->nchannels, obt->nchannels);
+ dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
+ dolog ("============================================\n");
+ dolog ("requested: buffer size %d period size %d\n",
+ req->buffer_size, req->period_size);
+ dolog ("obtained: buffer size %ld\n", obt->buffer_size);
+}
+#endif
+
+static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
+{
+ int err;
+ snd_pcm_sw_params_t *sw_params;
+
+ snd_pcm_sw_params_alloca (&sw_params);
+
+ err = snd_pcm_sw_params_current (handle, sw_params);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to get current software parameters\n");
+ return;
+ }
+
+ err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software threshold to %ld\n",
+ threshold);
+ return;
+ }
+
+ err = snd_pcm_sw_params (handle, sw_params);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software parameters\n");
+ return;
+ }
+}
+
+static int alsa_open (int in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep)
+{
+ snd_pcm_t *handle;
+ snd_pcm_hw_params_t *hw_params;
+ int err, freq, nchannels;
+ const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+ unsigned int period_size, buffer_size;
+ snd_pcm_uframes_t obt_buffer_size;
+ const char *typ = in ? "ADC" : "DAC";
+
+ freq = req->freq;
+ period_size = req->period_size;
+ buffer_size = req->buffer_size;
+ nchannels = req->nchannels;
+
+ snd_pcm_hw_params_alloca (&hw_params);
+
+ err = snd_pcm_open (
+ &handle,
+ pcm_name,
+ in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
+ return -1;
+ }
+
+ err = snd_pcm_hw_params_any (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_access (
+ handle,
+ hw_params,
+ SND_PCM_ACCESS_RW_INTERLEAVED
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set access type\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_channels_near (
+ handle,
+ hw_params,
+ &nchannels
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
+ req->nchannels);
+ goto err;
+ }
+
+ if (nchannels != 1 && nchannels != 2) {
+ alsa_logerr2 (err, typ,
+ "Can not handle obtained number of channels %d\n",
+ nchannels);
+ goto err;
+ }
+
+ if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
+ if (!buffer_size) {
+ buffer_size = DEFAULT_BUFFER_SIZE;
+ period_size= DEFAULT_PERIOD_SIZE;
+ }
+ }
+
+ if (buffer_size) {
+ if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
+ if (period_size) {
+ err = snd_pcm_hw_params_set_period_time_near (
+ handle,
+ hw_params,
+ &period_size,
+ 0);
+ if (err < 0) {
+ alsa_logerr2 (err, typ,
+ "Failed to set period time %d\n",
+ req->period_size);
+ goto err;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_buffer_time_near (
+ handle,
+ hw_params,
+ &buffer_size,
+ 0);
+
+ if (err < 0) {
+ alsa_logerr2 (err, typ,
+ "Failed to set buffer time %d\n",
+ req->buffer_size);
+ goto err;
+ }
+ }
+ else {
+ int dir;
+ snd_pcm_uframes_t minval;
+
+ if (period_size) {
+ minval = period_size;
+ dir = 0;
+
+ err = snd_pcm_hw_params_get_period_size_min (
+ hw_params,
+ &minval,
+ &dir
+ );
+ if (err < 0) {
+ alsa_logerr (
+ err,
+ "Can not get minmal period size for %s\n",
+ typ
+ );
+ }
+ else {
+ if (period_size < minval) {
+ if ((in && conf.period_size_in_overriden)
+ || (!in && conf.period_size_out_overriden)) {
+ dolog ("%s period size(%d) is less "
+ "than minmal period size(%ld)\n",
+ typ,
+ period_size,
+ minval);
+ }
+ period_size = minval;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_period_size (
+ handle,
+ hw_params,
+ period_size,
+ 0
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set period size %d\n",
+ req->period_size);
+ goto err;
+ }
+ }
+
+ minval = buffer_size;
+ err = snd_pcm_hw_params_get_buffer_size_min (
+ hw_params,
+ &minval
+ );
+ if (err < 0) {
+ alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+ typ);
+ }
+ else {
+ if (buffer_size < minval) {
+ if ((in && conf.buffer_size_in_overriden)
+ || (!in && conf.buffer_size_out_overriden)) {
+ dolog (
+ "%s buffer size(%d) is less "
+ "than minimal buffer size(%ld)\n",
+ typ,
+ buffer_size,
+ minval
+ );
+ }
+ buffer_size = minval;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_buffer_size (
+ handle,
+ hw_params,
+ buffer_size
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
+ req->buffer_size);
+ goto err;
+ }
+ }
+ }
+ else {
+ dolog ("warning: buffer size is not set\n");
+ }
+
+ err = snd_pcm_hw_params (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to get buffer size\n");
+ goto err;
+ }
+
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+ goto err;
+ }
+
+ obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
+ if (obt->can_pause < 0) {
+ alsa_logerr (err, "Can not get pause capability for %s\n", typ);
+ obt->can_pause = 0;
+ }
+
+ if (!in && conf.threshold) {
+ snd_pcm_uframes_t threshold;
+ int bytes_per_sec;
+
+ bytes_per_sec = freq
+ << (nchannels == 2)
+ << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+
+ threshold = (conf.threshold * bytes_per_sec) / 1000;
+ alsa_set_threshold (handle, threshold);
+ }
+
+ obt->fmt = req->fmt;
+ obt->nchannels = nchannels;
+ obt->freq = freq;
+ obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+ *handlep = handle;
+
+ if (obt->fmt != req->fmt ||
+ obt->nchannels != req->nchannels ||
+ obt->freq != req->freq) {
+#ifdef DEBUG_MISMATCHES
+ dolog ("Audio paramters mismatch for %s\n", typ);
+ alsa_dump_info (req, obt);
+#endif
+ }
+
+#ifdef DEBUG
+ alsa_dump_info (req, obt);
+#endif
+ return 0;
+
+ err:
+ alsa_anal_close (&handle);
+ return -1;
+}
+
+static int alsa_recover (snd_pcm_t *handle)
+{
+ int err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to prepare handle %p\n", handle);
+ return -1;
+ }
+ return 0;
+}
+
+static int alsa_run_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ int rpos, live, decr;
+ int samples;
+ uint8_t *dst;
+ st_sample_t *src;
+ snd_pcm_sframes_t avail;
+
+ live = audio_pcm_hw_get_live_out (hw);
+ if (!live) {
+ return 0;
+ }
+
+ avail = snd_pcm_avail_update (alsa->handle);
+ if (avail < 0) {
+ if (avail == -EPIPE) {
+ if (!alsa_recover (alsa->handle)) {
+ avail = snd_pcm_avail_update (alsa->handle);
+ if (avail >= 0) {
+ goto ok;
+ }
+ }
+ }
+
+ alsa_logerr (avail, "Can not get amount free space\n");
+ return 0;
+ }
+
+ ok:
+ decr = audio_MIN (live, avail);
+ samples = decr;
+ rpos = hw->rpos;
+ while (samples) {
+ int left_till_end_samples = hw->samples - rpos;
+ int convert_samples = audio_MIN (samples, left_till_end_samples);
+ snd_pcm_sframes_t written;
+
+ src = hw->mix_buf + rpos;
+ dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
+
+ hw->clip (dst, src, convert_samples);
+
+ again:
+ written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+
+ if (written < 0) {
+ switch (written) {
+ case -EPIPE:
+ if (!alsa_recover (alsa->handle)) {
+ goto again;
+ }
+ dolog (
+ "Failed to write %d frames to %p, handle %p not prepared\n",
+ convert_samples,
+ dst,
+ alsa->handle
+ );
+ goto exit;
+
+ case -EAGAIN:
+ goto again;
+
+ default:
+ alsa_logerr (written, "Failed to write %d frames to %p\n",
+ convert_samples, dst);
+ goto exit;
+ }
+ }
+
+ mixeng_clear (src, written);
+ rpos = (rpos + written) % hw->samples;
+ samples -= written;
+ }
+
+ exit:
+ hw->rpos = rpos;
+ return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ ldebug ("alsa_fini\n");
+ alsa_anal_close (&alsa->handle);
+
+ if (alsa->pcm_buf) {
+ qemu_free (alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+ }
+}
+
+static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ struct alsa_params_req req;
+ struct alsa_params_obt obt;
+ audfmt_e effective_fmt;
+ int endianness;
+ int err;
+ snd_pcm_t *handle;
+
+ req.fmt = aud_to_alsafmt (fmt);
+ req.freq = freq;
+ req.nchannels = nchannels;
+ req.period_size = conf.period_size_out;
+ req.buffer_size = conf.buffer_size_out;
+
+ if (alsa_open (0, &req, &obt, &handle)) {
+ return -1;
+ }
+
+ err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
+ alsa->can_pause = obt.can_pause;
+ hw->bufsize = obt.buffer_size;
+
+ alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ if (!alsa->pcm_buf) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ alsa->was_enabled = 0;
+ return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+ int err;
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ ldebug ("enabling voice\n");
+ audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
+ if (alsa->can_pause) {
+ /* Why this was_enabled madness is needed at all?? */
+ if (alsa->was_enabled) {
+ err = snd_pcm_pause (alsa->handle, 0);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to resume playing\n");
+ /* not fatal really */
+ }
+ }
+ else {
+ alsa->was_enabled = 1;
+ }
+ }
+ break;
+
+ case VOICE_DISABLE:
+ ldebug ("disabling voice\n");
+ if (alsa->can_pause) {
+ err = snd_pcm_pause (alsa->handle, 1);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to stop playing\n");
+ /* not fatal really */
+ }
+ }
+ break;
+ }
+ return 0;
+}
+
+static int alsa_init_in (HWVoiceIn *hw,
+ int freq, int nchannels, audfmt_e fmt)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ struct alsa_params_req req;
+ struct alsa_params_obt obt;
+ int endianness;
+ int err;
+ audfmt_e effective_fmt;
+ snd_pcm_t *handle;
+
+ req.fmt = aud_to_alsafmt (fmt);
+ req.freq = freq;
+ req.nchannels = nchannels;
+ req.period_size = conf.period_size_in;
+ req.buffer_size = conf.buffer_size_in;
+
+ if (alsa_open (1, &req, &obt, &handle)) {
+ return -1;
+ }
+
+ err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
+ alsa->can_pause = obt.can_pause;
+ hw->bufsize = obt.buffer_size;
+ alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ if (!alsa->pcm_buf) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ alsa_anal_close (&alsa->handle);
+
+ if (alsa->pcm_buf) {
+ qemu_free (alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+ }
+}
+
+static int alsa_run_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ int hwshift = hw->info.shift;
+ int i;
+ int live = audio_pcm_hw_get_live_in (hw);
+ int dead = hw->samples - live;
+ struct {
+ int add;
+ int len;
+ } bufs[2] = {
+ { hw->wpos, 0 },
+ { 0, 0 }
+ };
+
+ snd_pcm_uframes_t read_samples = 0;
+
+ if (!dead) {
+ return 0;
+ }
+
+ if (hw->wpos + dead > hw->samples) {
+ bufs[0].len = (hw->samples - hw->wpos);
+ bufs[1].len = (dead - (hw->samples - hw->wpos));
+ }
+ else {
+ bufs[0].len = dead;
+ }
+
+
+ for (i = 0; i < 2; ++i) {
+ void *src;
+ st_sample_t *dst;
+ snd_pcm_sframes_t nread;
+ snd_pcm_uframes_t len;
+
+ len = bufs[i].len;
+
+ src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
+ dst = hw->conv_buf + bufs[i].add;
+
+ while (len) {
+ nread = snd_pcm_readi (alsa->handle, src, len);
+
+ if (nread < 0) {
+ switch (nread) {
+ case -EPIPE:
+ if (!alsa_recover (alsa->handle)) {
+ continue;
+ }
+ dolog (
+ "Failed to read %ld frames from %p, "
+ "handle %p not prepared\n",
+ len,
+ src,
+ alsa->handle
+ );
+ goto exit;
+
+ case -EAGAIN:
+ continue;
+
+ default:
+ alsa_logerr (
+ nread,
+ "Failed to read %ld frames from %p\n",
+ len,
+ src
+ );
+ goto exit;
+ }
+ }
+
+ hw->conv (dst, src, nread, &nominal_volume);
+
+ src = advance (src, nread << hwshift);
+ dst += nread;
+
+ read_samples += nread;
+ len -= nread;
+ }
+ }
+
+ exit:
+ hw->wpos = (hw->wpos + read_samples) % hw->samples;
+ return read_samples;
+}
+
+static int alsa_read (SWVoiceIn *sw, void *buf, int size)
+{
+ return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ (void) hw;
+ (void) cmd;
+ return 0;
+}
+
+static void *alsa_audio_init (void)
+{
+ return &conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+ (void) opaque;
+}
+
+static struct audio_option alsa_options[] = {
+ {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
+ "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+ {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
+ "DAC period size", &conf.period_size_out_overriden, 0},
+ {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+ "DAC buffer size", &conf.buffer_size_out_overriden, 0},
+
+ {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
+ "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+ {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
+ "ADC period size", &conf.period_size_in_overriden, 0},
+ {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+ "ADC buffer size", &conf.buffer_size_in_overriden, 0},
+
+ {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+ "(undocumented)", NULL, 0},
+
+ {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
+ "DAC device name (for instance dmix)", NULL, 0},
+
+ {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
+ "ADC device name", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+ alsa_init_out,
+ alsa_fini_out,
+ alsa_run_out,
+ alsa_write,
+ alsa_ctl_out,
+
+ alsa_init_in,
+ alsa_fini_in,
+ alsa_run_in,
+ alsa_read,
+ alsa_ctl_in
+};
+
+struct audio_driver alsa_audio_driver = {
+ INIT_FIELD (name = ) "alsa",
+ INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
+ INIT_FIELD (options = ) alsa_options,
+ INIT_FIELD (init = ) alsa_audio_init,
+ INIT_FIELD (fini = ) alsa_audio_fini,
+ INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) INT_MAX,
+ INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
+};