audio merge (malc)
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1601 c046a42c-6fe2-441c-8c8c-71466251a162
diff --git a/Makefile.target b/Makefile.target
index cefad74..a22b2e0 100644
--- a/Makefile.target
+++ b/Makefile.target
@@ -453,8 +453,8 @@
include .depend
endif
-ifeq (0, 1)
+ifeq (1, 0)
audio.o sdlaudio.o dsoundaudio.o ossaudio.o wavaudio.o noaudio.o \
-fmodaudio.o alsaaudio.o mixeng.o: \
+fmodaudio.o alsaaudio.o mixeng.o sb16.o es1370.o gus.o adlib.o: \
CFLAGS := $(CFLAGS) -Wall -Werror -W -Wsign-compare
endif
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 1336905..65a0a0d 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -98,7 +98,7 @@
audfmt_e fmt;
int nchannels;
int can_pause;
- snd_pcm_uframes_t buffer_size;
+ snd_pcm_uframes_t samples;
};
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
@@ -121,7 +121,7 @@
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
@@ -209,7 +209,7 @@
return 0;
}
-#ifdef DEBUG_MISMATCHES
+#if defined DEBUG_MISMATCHES || defined DEBUG
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt)
{
@@ -221,7 +221,7 @@
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);
+ dolog ("obtained: samples %ld\n", obt->samples);
}
#endif
@@ -234,14 +234,14 @@
err = snd_pcm_sw_params_current (handle, sw_params);
if (err < 0) {
- dolog ("Can not fully initialize DAC\n");
+ 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 ("Can not fully initialize DAC\n");
+ dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software threshold to %ld\n",
threshold);
return;
@@ -249,7 +249,7 @@
err = snd_pcm_sw_params (handle, sw_params);
if (err < 0) {
- dolog ("Can not fully initialize DAC\n");
+ dolog ("Could not fully initialize DAC\n");
alsa_logerr (err, "Failed to set software parameters\n");
return;
}
@@ -344,7 +344,8 @@
handle,
hw_params,
&period_size,
- 0);
+ 0
+ );
if (err < 0) {
alsa_logerr2 (err, typ,
"Failed to set period time %d\n",
@@ -357,7 +358,8 @@
handle,
hw_params,
&buffer_size,
- 0);
+ 0
+ );
if (err < 0) {
alsa_logerr2 (err, typ,
@@ -382,7 +384,7 @@
if (err < 0) {
alsa_logerr (
err,
- "Can not get minmal period size for %s\n",
+ "Could not get minmal period size for %s\n",
typ
);
}
@@ -419,7 +421,7 @@
&minval
);
if (err < 0) {
- alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+ alsa_logerr (err, "Could not get minmal buffer size for %s\n",
typ);
}
else {
@@ -451,7 +453,7 @@
}
}
else {
- dolog ("warning: buffer size is not set\n");
+ dolog ("warning: Buffer size is not set\n");
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -468,13 +470,13 @@
err = snd_pcm_prepare (handle);
if (err < 0) {
- alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+ alsa_logerr2 (err, typ, "Could 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);
+ alsa_logerr (err, "Could not get pause capability for %s\n", typ);
obt->can_pause = 0;
}
@@ -493,17 +495,17 @@
obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
- obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+ obt->samples = obt_buffer_size;
*handlep = handle;
+#if defined DEBUG_MISMATCHES || defined DEBUG
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
}
+#endif
#ifdef DEBUG
alsa_dump_info (req, obt);
@@ -550,7 +552,7 @@
}
}
- alsa_logerr (avail, "Can not get amount free space\n");
+ alsa_logerr (avail, "Could not get amount free space\n");
return 0;
}
@@ -618,7 +620,7 @@
}
}
-static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
@@ -627,10 +629,11 @@
int endianness;
int err;
snd_pcm_t *handle;
+ audsettings_t obt_as;
- req.fmt = aud_to_alsafmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_alsafmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.period_size = conf.period_size_out;
req.buffer_size = conf.buffer_size_out;
@@ -644,18 +647,22 @@
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
alsa->can_pause = obt.can_pause;
- hw->bufsize = obt.buffer_size;
+ hw->samples = obt.samples;
- alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
+ dolog ("Could not allocate DAC buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
alsa_anal_close (&handle);
return -1;
}
@@ -703,8 +710,7 @@
return 0;
}
-static int alsa_init_in (HWVoiceIn *hw,
- int freq, int nchannels, audfmt_e fmt)
+static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
@@ -713,10 +719,11 @@
int err;
audfmt_e effective_fmt;
snd_pcm_t *handle;
+ audsettings_t obt_as;
- req.fmt = aud_to_alsafmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_alsafmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.period_size = conf.period_size_in;
req.buffer_size = conf.buffer_size_in;
@@ -730,17 +737,22 @@
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
alsa->can_pause = obt.can_pause;
- hw->bufsize = obt.buffer_size;
- alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ 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 bytes)\n",
+ hw->samples << hw->info.shift);
alsa_anal_close (&handle);
return -1;
}
diff --git a/audio/audio.c b/audio/audio.c
index 1a3925d..961654b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -26,16 +26,12 @@
#define AUDIO_CAP "audio"
#include "audio_int.h"
-static void audio_pcm_hw_fini_in (HWVoiceIn *hw);
-static void audio_pcm_hw_fini_out (HWVoiceOut *hw);
-
-static LIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
-static LIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
-
/* #define DEBUG_PLIVE */
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
+#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
+
static struct audio_driver *drvtab[] = {
#ifdef CONFIG_OSS
&oss_audio_driver,
@@ -59,31 +55,50 @@
&wav_audio_driver
};
-AudioState audio_state = {
- /* Out */
- 1, /* use fixed settings */
- 44100, /* fixed frequency */
- 2, /* fixed channels */
- AUD_FMT_S16, /* fixed format */
- 1, /* number of hw voices */
- 1, /* greedy */
+struct fixed_settings {
+ int enabled;
+ int nb_voices;
+ int greedy;
+ audsettings_t settings;
+};
- /* In */
- 1, /* use fixed settings */
- 44100, /* fixed frequency */
- 2, /* fixed channels */
- AUD_FMT_S16, /* fixed format */
- 1, /* number of hw voices */
- 1, /* greedy */
+static struct {
+ struct fixed_settings fixed_out;
+ struct fixed_settings fixed_in;
+ union {
+ int hz;
+ int64_t ticks;
+ } period;
+ int plive;
+} conf = {
+ { /* DAC fixed settings */
+ 1, /* enabled */
+ 1, /* nb_voices */
+ 1, /* greedy */
+ {
+ 44100, /* freq */
+ 2, /* nchannels */
+ AUD_FMT_S16 /* fmt */
+ }
+ },
- NULL, /* driver opaque */
- NULL, /* driver */
+ { /* ADC fixed settings */
+ 1, /* enabled */
+ 1, /* nb_voices */
+ 1, /* greedy */
+ {
+ 44100, /* freq */
+ 2, /* nchannels */
+ AUD_FMT_S16 /* fmt */
+ }
+ },
- NULL, /* timer handle */
{ 0 }, /* period */
0 /* plive */
};
+static AudioState glob_audio_state;
+
volume_t nominal_volume = {
0,
#ifdef FLOAT_MIXENG
@@ -148,6 +163,26 @@
}
#endif
+void *audio_calloc (const char *funcname, int nmemb, size_t size)
+{
+ int cond;
+ size_t len;
+
+ len = nmemb * size;
+ cond = !nmemb || !size;
+ cond |= nmemb < 0;
+ cond |= len < size;
+
+ if (audio_bug ("audio_calloc", cond)) {
+ AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
+ funcname);
+ AUD_log (NULL, "nmemb=%d size=%d (len=%d)\n", nmemb, size, len);
+ return NULL;
+ }
+
+ return qemu_mallocz (len);
+}
+
static char *audio_alloc_prefix (const char *s)
{
const char qemu_prefix[] = "QEMU_";
@@ -386,14 +421,19 @@
}
len = strlen (opt->name);
+ /* len of opt->name + len of prefix + size of qemu_prefix
+ * (includes trailing zero) + zero + underscore (on behalf of
+ * sizeof) */
optname = qemu_malloc (len + preflen + sizeof (qemu_prefix) + 1);
if (!optname) {
- dolog ("Can not allocate memory for option name `%s'\n",
+ dolog ("Could not allocate memory for option name `%s'\n",
opt->name);
continue;
}
strcpy (optname, qemu_prefix);
+
+ /* copy while upper-casing, including trailing zero */
for (i = 0; i <= preflen; ++i) {
optname[i + sizeof (qemu_prefix) - 1] = toupper (prefix[i]);
}
@@ -438,12 +478,60 @@
}
}
-static int audio_pcm_info_eq (struct audio_pcm_info *info, int freq,
- int nchannels, audfmt_e fmt)
+static void audio_print_settings (audsettings_t *as)
+{
+ dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
+
+ switch (as->fmt) {
+ case AUD_FMT_S8:
+ AUD_log (NULL, "S8");
+ break;
+ case AUD_FMT_U8:
+ AUD_log (NULL, "U8");
+ break;
+ case AUD_FMT_S16:
+ AUD_log (NULL, "S16");
+ break;
+ case AUD_FMT_U16:
+ AUD_log (NULL, "U16");
+ break;
+ default:
+ AUD_log (NULL, "invalid(%d)", as->fmt);
+ break;
+ }
+ AUD_log (NULL, "\n");
+}
+
+static int audio_validate_settigs (audsettings_t *as)
+{
+ int invalid;
+
+ invalid = as->nchannels != 1 && as->nchannels != 2;
+
+ switch (as->fmt) {
+ case AUD_FMT_S8:
+ case AUD_FMT_U8:
+ case AUD_FMT_S16:
+ case AUD_FMT_U16:
+ break;
+ default:
+ invalid = 1;
+ break;
+ }
+
+ invalid |= as->freq <= 0;
+
+ if (invalid) {
+ return -1;
+ }
+ return 0;
+}
+
+static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
{
int bits = 8, sign = 0;
- switch (fmt) {
+ switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
case AUD_FMT_U8:
@@ -455,18 +543,21 @@
bits = 16;
break;
}
- return info->freq == freq
- && info->nchannels == nchannels
+ return info->freq == as->freq
+ && info->nchannels == as->nchannels
&& info->sign == sign
&& info->bits == bits;
}
-void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
- int nchannels, audfmt_e fmt, int swap_endian)
+void audio_pcm_init_info (
+ struct audio_pcm_info *info,
+ audsettings_t *as,
+ int swap_endian
+ )
{
int bits = 8, sign = 0;
- switch (fmt) {
+ switch (as->fmt) {
case AUD_FMT_S8:
sign = 1;
case AUD_FMT_U8:
@@ -479,11 +570,11 @@
break;
}
- info->freq = freq;
+ info->freq = as->freq;
info->bits = bits;
info->sign = sign;
- info->nchannels = nchannels;
- info->shift = (nchannels == 2) + (bits == 16);
+ info->nchannels = as->nchannels;
+ info->shift = (as->nchannels == 2) + (bits == 16);
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->swap_endian = swap_endian;
@@ -532,38 +623,16 @@
static int audio_pcm_hw_alloc_resources_in (HWVoiceIn *hw)
{
- hw->conv_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+ hw->conv_buf = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (st_sample_t));
if (!hw->conv_buf) {
+ dolog ("Could not allocate ADC conversion buffer (%d bytes)\n",
+ hw->samples * sizeof (st_sample_t));
return -1;
}
return 0;
}
-static int audio_pcm_hw_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
-{
- audio_pcm_hw_fini_in (hw);
-
- if (hw->pcm_ops->init_in (hw, freq, nchannels, fmt)) {
- memset (hw, 0, audio_state.drv->voice_size_in);
- return -1;
- }
- LIST_INIT (&hw->sw_head);
- hw->active = 1;
- hw->samples = hw->bufsize >> hw->info.shift;
- hw->conv =
- mixeng_conv
- [nchannels == 2]
- [hw->info.sign]
- [hw->info.swap_endian]
- [hw->info.bits == 16];
- if (audio_pcm_hw_alloc_resources_in (hw)) {
- audio_pcm_hw_free_resources_in (hw);
- return -1;
- }
- return 0;
-}
-
-static uint64_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
int m = hw->total_samples_captured;
@@ -606,8 +675,10 @@
static int audio_pcm_sw_alloc_resources_in (SWVoiceIn *sw)
{
int samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
- sw->conv_buf = qemu_mallocz (samples * sizeof (st_sample_t));
+ sw->conv_buf = audio_calloc (AUDIO_FUNC, samples, sizeof (st_sample_t));
if (!sw->conv_buf) {
+ dolog ("Could not allocate buffer for `%s' (%d bytes)\n",
+ SW_NAME (sw), samples * sizeof (st_sample_t));
return -1;
}
@@ -620,19 +691,22 @@
return 0;
}
-static int audio_pcm_sw_init_in (SWVoiceIn *sw, HWVoiceIn *hw, const char *name,
- int freq, int nchannels, audfmt_e fmt)
+static int audio_pcm_sw_init_in (
+ SWVoiceIn *sw,
+ HWVoiceIn *hw,
+ const char *name,
+ audsettings_t *as
+ )
{
- audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
- /* None of the cards emulated by QEMU are big-endian
- hence following shortcut */
- audio_need_to_swap_endian (0));
+ /* None of the cards emulated by QEMU are big-endian
+ hence following shortcut */
+ audio_pcm_init_info (&sw->info, as, audio_need_to_swap_endian (0));
sw->hw = hw;
sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
sw->clip =
mixeng_clip
- [nchannels == 2]
+ [sw->info.nchannels == 2]
[sw->info.sign]
[sw->info.swap_endian]
[sw->info.bits == 16];
@@ -699,6 +773,7 @@
if (audio_bug (AUDIO_FUNC, osamp < 0)) {
dolog ("osamp=%d\n", osamp);
+ return 0;
}
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
@@ -717,6 +792,27 @@
/*
* Hard voice (playback)
*/
+static void audio_pcm_hw_free_resources_out (HWVoiceOut *hw)
+{
+ if (hw->mix_buf) {
+ qemu_free (hw->mix_buf);
+ }
+
+ hw->mix_buf = NULL;
+}
+
+static int audio_pcm_hw_alloc_resources_out (HWVoiceOut *hw)
+{
+ hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (st_sample_t));
+ if (!hw->mix_buf) {
+ dolog ("Could not allocate DAC mixing buffer (%d bytes)\n",
+ hw->samples * sizeof (st_sample_t));
+ return -1;
+ }
+
+ return 0;
+}
+
static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
@@ -734,50 +830,6 @@
return m;
}
-static void audio_pcm_hw_free_resources_out (HWVoiceOut *hw)
-{
- if (hw->mix_buf) {
- qemu_free (hw->mix_buf);
- }
-
- hw->mix_buf = NULL;
-}
-
-static int audio_pcm_hw_alloc_resources_out (HWVoiceOut *hw)
-{
- hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
- if (!hw->mix_buf) {
- return -1;
- }
-
- return 0;
-}
-
-static int audio_pcm_hw_init_out (HWVoiceOut *hw, int freq,
- int nchannels, audfmt_e fmt)
-{
- audio_pcm_hw_fini_out (hw);
- if (hw->pcm_ops->init_out (hw, freq, nchannels, fmt)) {
- memset (hw, 0, audio_state.drv->voice_size_out);
- return -1;
- }
-
- LIST_INIT (&hw->sw_head);
- hw->active = 1;
- hw->samples = hw->bufsize >> hw->info.shift;
- hw->clip =
- mixeng_clip
- [nchannels == 2]
- [hw->info.sign]
- [hw->info.swap_endian]
- [hw->info.bits == 16];
- if (audio_pcm_hw_alloc_resources_out (hw)) {
- audio_pcm_hw_fini_out (hw);
- return -1;
- }
- return 0;
-}
-
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
int smin;
@@ -830,8 +882,10 @@
static int audio_pcm_sw_alloc_resources_out (SWVoiceOut *sw)
{
- sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
+ sw->buf = audio_calloc (AUDIO_FUNC, sw->hw->samples, sizeof (st_sample_t));
if (!sw->buf) {
+ dolog ("Could not allocate buffer for `%s' (%d bytes)\n",
+ SW_NAME (sw), sw->hw->samples * sizeof (st_sample_t));
return -1;
}
@@ -844,14 +898,16 @@
return 0;
}
-static int audio_pcm_sw_init_out (SWVoiceOut *sw, HWVoiceOut *hw,
- const char *name, int freq,
- int nchannels, audfmt_e fmt)
+static int audio_pcm_sw_init_out (
+ SWVoiceOut *sw,
+ HWVoiceOut *hw,
+ const char *name,
+ audsettings_t *as
+ )
{
- audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
- /* None of the cards emulated by QEMU are big-endian
- hence following shortcut */
- audio_need_to_swap_endian (0));
+ /* None of the cards emulated by QEMU are big-endian
+ hence following shortcut */
+ audio_pcm_init_info (&sw->info, as, audio_need_to_swap_endian (0));
sw->hw = hw;
sw->empty = 1;
sw->active = 0;
@@ -860,7 +916,7 @@
sw->conv =
mixeng_conv
- [nchannels == 2]
+ [sw->info.nchannels == 2]
[sw->info.sign]
[sw->info.swap_endian]
[sw->info.bits == 16];
@@ -930,12 +986,11 @@
#ifdef DEBUG_OUT
dolog (
- "%s: write size %d ret %d total sw %d, hw %d\n",
- sw->name,
+ "%s: write size %d ret %d total sw %d\n",
+ SW_NAME (sw),
size >> sw->info.shift,
ret,
- sw->total_hw_samples_mixed,
- sw->hw->total_samples_played
+ sw->total_hw_samples_mixed
);
#endif
@@ -965,7 +1020,7 @@
}
if (!sw->hw->enabled) {
- dolog ("Writing to disabled voice %s\n", sw->name);
+ dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
@@ -983,7 +1038,7 @@
}
if (!sw->hw->enabled) {
- dolog ("Reading from disabled voice %s\n", sw->name);
+ dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
@@ -993,7 +1048,7 @@
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->bufsize;
+ return sw->hw->samples << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -1091,7 +1146,7 @@
ldebug (
"%s: get_avail live %d ret %lld\n",
- sw->name,
+ SW_NAME (sw),
live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
);
@@ -1110,34 +1165,37 @@
if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ return 0;
}
dead = sw->hw->samples - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %lld\n",
- sw->name,
+ SW_NAME (sw),
live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
#endif
return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}
-static void audio_run_out (void)
+static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
- while ((hw = audio_pcm_hw_find_any_active_enabled_out (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
int played;
- int live, free, nb_live;
+ int live, free, nb_live, cleanup_required;
live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
if (!nb_live) {
live = 0;
}
+
if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ continue;
}
if (hw->pending_disable && !nb_live) {
@@ -1170,15 +1228,15 @@
}
#ifdef DEBUG_OUT
- dolog ("played = %d total %d\n", played, hw->total_samples_played);
+ dolog ("played=%d\n", played);
#endif
if (played) {
hw->ts_helper += played;
}
+ cleanup_required = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
- again:
if (!sw->active && sw->empty) {
continue;
}
@@ -1193,22 +1251,7 @@
if (!sw->total_hw_samples_mixed) {
sw->empty = 1;
-
- if (!sw->active && !sw->callback.fn) {
- SWVoiceOut *temp = sw->entries.le_next;
-
-#ifdef DEBUG_PLIVE
- dolog ("Finishing with old voice\n");
-#endif
- AUD_close_out (sw);
- sw = temp;
- if (sw) {
- goto again;
- }
- else {
- break;
- }
- }
+ cleanup_required |= !sw->active && !sw->callback.fn;
}
if (sw->active) {
@@ -1218,14 +1261,27 @@
}
}
}
+
+ if (cleanup_required) {
+ restart:
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (!sw->active && !sw->callback.fn) {
+#ifdef DEBUG_PLIVE
+ dolog ("Finishing with old voice\n");
+#endif
+ audio_close_out (s, sw);
+ goto restart; /* play it safe */
+ }
+ }
+ }
}
}
-static void audio_run_in (void)
+static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
- while ((hw = audio_pcm_hw_find_any_active_enabled_in (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in (s, hw))) {
SWVoiceIn *sw;
int captured, min;
@@ -1252,42 +1308,42 @@
static struct audio_option audio_options[] = {
/* DAC */
- {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+ {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_out.enabled,
"Use fixed settings for host DAC", NULL, 0},
- {"DAC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
+ {"DAC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_out.settings.freq,
"Frequency for fixed host DAC", NULL, 0},
- {"DAC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
+ {"DAC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_out.settings.fmt,
"Format for fixed host DAC", NULL, 0},
- {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_out,
+ {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_out.settings.nchannels,
"Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},
- {"DAC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+ {"DAC_VOICES", AUD_OPT_INT, &conf.fixed_out.nb_voices,
"Number of voices for DAC", NULL, 0},
/* ADC */
- {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+ {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &conf.fixed_in.enabled,
"Use fixed settings for host ADC", NULL, 0},
- {"ADC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
- "Frequency for fixed ADC", NULL, 0},
+ {"ADC_FIXED_FREQ", AUD_OPT_INT, &conf.fixed_in.settings.freq,
+ "Frequency for fixed host ADC", NULL, 0},
- {"ADC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
- "Format for fixed ADC", NULL, 0},
+ {"ADC_FIXED_FMT", AUD_OPT_FMT, &conf.fixed_in.settings.fmt,
+ "Format for fixed host ADC", NULL, 0},
- {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_in,
+ {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &conf.fixed_in.settings.nchannels,
"Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},
- {"ADC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+ {"ADC_VOICES", AUD_OPT_INT, &conf.fixed_in.nb_voices,
"Number of voices for ADC", NULL, 0},
/* Misc */
- {"TIMER_PERIOD", AUD_OPT_INT, &audio_state.period.usec,
- "Timer period in microseconds (0 - try lowest possible)", NULL, 0},
+ {"TIMER_PERIOD", AUD_OPT_INT, &conf.period.hz,
+ "Timer period in HZ (0 - use lowest possible)", NULL, 0},
- {"PLIVE", AUD_OPT_BOOL, &audio_state.plive,
+ {"PLIVE", AUD_OPT_BOOL, &conf.plive,
"(undocumented)", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
@@ -1378,25 +1434,21 @@
{
AudioState *s = opaque;
- audio_run_out ();
- audio_run_in ();
+ audio_run_out (s);
+ audio_run_in (s);
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
+ qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
}
-static int audio_driver_init (struct audio_driver *drv)
+static int audio_driver_init (AudioState *s, struct audio_driver *drv)
{
if (drv->options) {
audio_process_options (drv->name, drv->options);
}
- audio_state.opaque = drv->init ();
+ s->drv_opaque = drv->init ();
- if (audio_state.opaque) {
- int i;
- HWVoiceOut *hwo;
- HWVoiceIn *hwi;
-
- if (audio_state.nb_hw_voices_out > drv->max_voices_out) {
+ if (s->drv_opaque) {
+ if (s->nb_hw_voices_out > drv->max_voices_out) {
if (!drv->max_voices_out) {
dolog ("`%s' does not support DAC\n", drv->name);
}
@@ -1405,30 +1457,13 @@
"`%s' does not support %d multiple DAC voicess\n"
"Resetting to %d\n",
drv->name,
- audio_state.nb_hw_voices_out,
+ s->nb_hw_voices_out,
drv->max_voices_out
);
}
- audio_state.nb_hw_voices_out = drv->max_voices_out;
+ s->nb_hw_voices_out = drv->max_voices_out;
}
- LIST_INIT (&hw_head_out);
- hwo = qemu_mallocz (audio_state.nb_hw_voices_out * drv->voice_size_out);
- if (!hwo) {
- dolog (
- "Not enough memory for %d `%s' DAC voices (each %d bytes)\n",
- audio_state.nb_hw_voices_out,
- drv->name,
- drv->voice_size_out
- );
- drv->fini (audio_state.opaque);
- return -1;
- }
-
- for (i = 0; i < audio_state.nb_hw_voices_out; ++i) {
- LIST_INSERT_HEAD (&hw_head_out, hwo, entries);
- hwo = advance (hwo, drv->voice_size_out);
- }
if (!drv->voice_size_in && drv->max_voices_in) {
ldebug ("warning: No ADC voice size defined for `%s'\n",
@@ -1442,16 +1477,16 @@
}
if (drv->voice_size_in && !drv->max_voices_in) {
- ldebug ("warning: ADC voice size is %d for ADC less driver `%s'\n",
- drv->voice_size_out, drv->name);
+ ldebug ("warning: `%s' ADC voice size %d, zero voices \n",
+ drv->name, drv->voice_size_out);
}
if (drv->voice_size_out && !drv->max_voices_out) {
- ldebug ("warning: DAC voice size is %d for DAC less driver `%s'\n",
- drv->voice_size_in, drv->name);
+ ldebug ("warning: `%s' DAC voice size %d, zero voices \n",
+ drv->name, drv->voice_size_in);
}
- if (audio_state.nb_hw_voices_in > drv->max_voices_in) {
+ if (s->nb_hw_voices_in > drv->max_voices_in) {
if (!drv->max_voices_in) {
ldebug ("`%s' does not support ADC\n", drv->name);
}
@@ -1460,33 +1495,16 @@
"`%s' does not support %d multiple ADC voices\n"
"Resetting to %d\n",
drv->name,
- audio_state.nb_hw_voices_in,
+ s->nb_hw_voices_in,
drv->max_voices_in
);
}
- audio_state.nb_hw_voices_in = drv->max_voices_in;
+ s->nb_hw_voices_in = drv->max_voices_in;
}
- LIST_INIT (&hw_head_in);
- hwi = qemu_mallocz (audio_state.nb_hw_voices_in * drv->voice_size_in);
- if (!hwi) {
- dolog (
- "Not enough memory for %d `%s' ADC voices (each %d bytes)\n",
- audio_state.nb_hw_voices_in,
- drv->name,
- drv->voice_size_in
- );
- qemu_free (hwo);
- drv->fini (audio_state.opaque);
- return -1;
- }
-
- for (i = 0; i < audio_state.nb_hw_voices_in; ++i) {
- LIST_INSERT_HEAD (&hw_head_in, hwi, entries);
- hwi = advance (hwi, drv->voice_size_in);
- }
-
- audio_state.drv = drv;
+ LIST_INIT (&s->hw_head_out);
+ LIST_INIT (&s->hw_head_in);
+ s->drv = drv;
return 0;
}
else {
@@ -1497,12 +1515,12 @@
static void audio_vm_stop_handler (void *opaque, int reason)
{
+ AudioState *s = opaque;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
int op = reason ? VOICE_ENABLE : VOICE_DISABLE;
- (void) opaque;
- while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_out (s, hwo))) {
if (!hwo->pcm_ops) {
continue;
}
@@ -1512,7 +1530,7 @@
}
}
- while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_in (s, hwi))) {
if (!hwi->pcm_ops) {
continue;
}
@@ -1525,10 +1543,11 @@
static void audio_atexit (void)
{
+ AudioState *s = &glob_audio_state;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
- while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_out (s, hwo))) {
if (!hwo->pcm_ops) {
continue;
}
@@ -1539,7 +1558,7 @@
hwo->pcm_ops->fini_out (hwo);
}
- while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_in (s, hwi))) {
if (!hwi->pcm_ops) {
continue;
}
@@ -1549,7 +1568,10 @@
}
hwi->pcm_ops->fini_in (hwi);
}
- audio_state.drv->fini (audio_state.opaque);
+
+ if (s->drv) {
+ s->drv->fini (s->drv_opaque);
+ }
}
static void audio_save (QEMUFile *f, void *opaque)
@@ -1570,15 +1592,33 @@
return 0;
}
-void AUD_init (void)
+void AUD_register_card (AudioState *s, const char *name, QEMUSoundCard *card)
+{
+ card->audio = s;
+ card->name = qemu_strdup (name);
+ memset (&card->entries, 0, sizeof (card->entries));
+ LIST_INSERT_HEAD (&s->card_head, card, entries);
+}
+
+void AUD_remove_card (QEMUSoundCard *card)
+{
+ LIST_REMOVE (card, entries);
+ card->audio = NULL;
+ qemu_free (card->name);
+}
+
+AudioState *AUD_init (void)
{
size_t i;
int done = 0;
const char *drvname;
- AudioState *s = &audio_state;
+ AudioState *s = &glob_audio_state;
audio_process_options ("AUDIO", audio_options);
+ s->nb_hw_voices_out = conf.fixed_out.nb_voices;
+ s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of DAC voices %d\n",
s->nb_hw_voices_out);
@@ -1598,8 +1638,8 @@
s->ts = qemu_new_timer (vm_clock, audio_timer, s);
if (!s->ts) {
- dolog ("Can not create audio timer\n");
- return;
+ dolog ("Could not create audio timer\n");
+ return NULL;
}
if (drvname) {
@@ -1607,7 +1647,7 @@
for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
if (!strcmp (drvname, drvtab[i]->name)) {
- done = !audio_driver_init (drvtab[i]);
+ done = !audio_driver_init (s, drvtab[i]);
found = 1;
break;
}
@@ -1619,37 +1659,47 @@
}
}
- qemu_add_vm_stop_handler (audio_vm_stop_handler, NULL);
- atexit (audio_atexit);
-
if (!done) {
for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
if (drvtab[i]->can_be_default) {
- done = !audio_driver_init (drvtab[i]);
+ done = !audio_driver_init (s, drvtab[i]);
}
}
}
- register_savevm ("audio", 0, 1, audio_save, audio_load, NULL);
if (!done) {
- if (audio_driver_init (&no_audio_driver)) {
- dolog ("Can not initialize audio subsystem\n");
+ done = !audio_driver_init (s, &no_audio_driver);
+ if (!done) {
+ dolog ("Could not initialize audio subsystem\n");
}
else {
- dolog ("warning: using timer based audio emulation\n");
+ dolog ("warning: Using timer based audio emulation\n");
}
}
- if (s->period.usec <= 0) {
- if (s->period.usec < 0) {
- dolog ("warning: timer period is negative - %d treating as zero\n",
- s->period.usec);
+ if (done) {
+ if (conf.period.hz <= 0) {
+ if (conf.period.hz < 0) {
+ dolog ("warning: Timer period is negative - %d "
+ "treating as zero\n",
+ conf.period.hz);
+ }
+ conf.period.ticks = 1;
}
- s->period.ticks = 1;
+ else {
+ conf.period.ticks = ticks_per_sec / conf.period.hz;
+ }
+
+ qemu_add_vm_stop_handler (audio_vm_stop_handler, NULL);
}
else {
- s->period.ticks = (ticks_per_sec * s->period.usec) / 1000000;
+ qemu_del_timer (s->ts);
+ return NULL;
}
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
+ LIST_INIT (&s->card_head);
+ register_savevm ("audio", 0, 1, audio_save, audio_load, s);
+ atexit (audio_atexit);
+ qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
+ return s;
}
diff --git a/audio/audio.h b/audio/audio.h
index 6dd2fd2..682d0e0 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -24,18 +24,33 @@
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
+#include "sys-queue.h"
+
typedef void (*audio_callback_fn_t) (void *opaque, int avail);
typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16
+ AUD_FMT_U8,
+ AUD_FMT_S8,
+ AUD_FMT_U16,
+ AUD_FMT_S16
} audfmt_e;
+typedef struct {
+ int freq;
+ int nchannels;
+ audfmt_e fmt;
+} audsettings_t;
+
+typedef struct AudioState AudioState;
typedef struct SWVoiceOut SWVoiceOut;
typedef struct SWVoiceIn SWVoiceIn;
+typedef struct QEMUSoundCard {
+ AudioState *audio;
+ char *name;
+ LIST_ENTRY (QEMUSoundCard) entries;
+} QEMUSoundCard;
+
typedef struct QEMUAudioTimeStamp {
uint64_t old_ts;
} QEMUAudioTimeStamp;
@@ -47,46 +62,45 @@
#endif
;
-void AUD_init (void);
+AudioState *AUD_init (void);
void AUD_help (void);
+void AUD_register_card (AudioState *s, const char *name, QEMUSoundCard *card);
+void AUD_remove_card (QEMUSoundCard *card);
-SWVoiceOut *AUD_open_out (
+SWVoiceOut *AUD_open_out (
+ QEMUSoundCard *card,
SWVoiceOut *sw,
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- int freq,
- int nchannels,
- audfmt_e fmt
+ audsettings_t *settings
);
-void AUD_close_out (SWVoiceOut *sw);
-int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
-int AUD_get_buffer_size_out (SWVoiceOut *sw);
-void AUD_set_active_out (SWVoiceOut *sw, int on);
-int AUD_is_active_out (SWVoiceOut *sw);
-void AUD_init_time_stamp_out (SWVoiceOut *sw,
- QEMUAudioTimeStamp *ts);
-uint64_t AUD_time_stamp_get_elapsed_usec_out (SWVoiceOut *sw,
- QEMUAudioTimeStamp *ts);
-SWVoiceIn *AUD_open_in (
+void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
+int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+int AUD_get_buffer_size_out (SWVoiceOut *sw);
+void AUD_set_active_out (SWVoiceOut *sw, int on);
+int AUD_is_active_out (SWVoiceOut *sw);
+
+void AUD_init_time_stamp_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
+uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
+
+SWVoiceIn *AUD_open_in (
+ QEMUSoundCard *card,
SWVoiceIn *sw,
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- int freq,
- int nchannels,
- audfmt_e fmt
+ audsettings_t *settings
);
-void AUD_close_in (SWVoiceIn *sw);
-int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
-void AUD_adjust_in (SWVoiceIn *sw, int leftover);
-void AUD_set_active_in (SWVoiceIn *sw, int on);
-int AUD_is_active_in (SWVoiceIn *sw);
-void AUD_init_time_stamp_in (SWVoiceIn *sw,
- QEMUAudioTimeStamp *ts);
-uint64_t AUD_time_stamp_get_elapsed_usec_in (SWVoiceIn *sw,
- QEMUAudioTimeStamp *ts);
+
+void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
+int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+void AUD_set_active_in (SWVoiceIn *sw, int on);
+int AUD_is_active_in (SWVoiceIn *sw);
+
+void AUD_init_time_stamp_in (SWVoiceIn *sw, QEMUAudioTimeStamp *ts);
+uint64_t AUD_get_elapsed_usec_in (SWVoiceIn *sw, QEMUAudioTimeStamp *ts);
static inline void *advance (void *p, int incr)
{
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9d28829..6d4c32b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -24,16 +24,12 @@
#ifndef QEMU_AUDIO_INT_H
#define QEMU_AUDIO_INT_H
-#include "sys-queue.h"
-
#ifdef CONFIG_COREAUDIO
#define FLOAT_MIXENG
/* #define RECIPROCAL */
#endif
#include "mixeng.h"
-int audio_bug (const char *funcname, int cond);
-
struct audio_pcm_ops;
typedef enum {
@@ -69,7 +65,6 @@
};
typedef struct HWVoiceOut {
- int active;
int enabled;
int pending_disable;
int valid;
@@ -78,7 +73,6 @@
f_sample *clip;
int rpos;
- int bufsize;
uint64_t ts_helper;
st_sample_t *mix_buf;
@@ -91,13 +85,11 @@
typedef struct HWVoiceIn {
int enabled;
- int active;
struct audio_pcm_info info;
t_sample *conv;
int wpos;
- int bufsize;
int total_samples_captured;
uint64_t ts_helper;
@@ -109,58 +101,6 @@
LIST_ENTRY (HWVoiceIn) entries;
} HWVoiceIn;
-extern struct audio_driver no_audio_driver;
-extern struct audio_driver oss_audio_driver;
-extern struct audio_driver sdl_audio_driver;
-extern struct audio_driver wav_audio_driver;
-extern struct audio_driver fmod_audio_driver;
-extern struct audio_driver alsa_audio_driver;
-extern struct audio_driver coreaudio_audio_driver;
-extern struct audio_driver dsound_audio_driver;
-extern volume_t nominal_volume;
-
-struct audio_driver {
- const char *name;
- const char *descr;
- struct audio_option *options;
- void *(*init) (void);
- void (*fini) (void *);
- struct audio_pcm_ops *pcm_ops;
- int can_be_default;
- int max_voices_out;
- int max_voices_in;
- int voice_size_out;
- int voice_size_in;
-};
-
-typedef struct AudioState {
- int fixed_settings_out;
- int fixed_freq_out;
- int fixed_channels_out;
- int fixed_fmt_out;
- int nb_hw_voices_out;
- int greedy_out;
-
- int fixed_settings_in;
- int fixed_freq_in;
- int fixed_channels_in;
- int fixed_fmt_in;
- int nb_hw_voices_in;
- int greedy_in;
-
- void *opaque;
- struct audio_driver *drv;
-
- QEMUTimer *ts;
- union {
- int usec;
- int64_t ticks;
- } period;
-
- int plive;
-} AudioState;
-extern AudioState audio_state;
-
struct SWVoiceOut {
struct audio_pcm_info info;
t_sample *conv;
@@ -192,22 +132,58 @@
LIST_ENTRY (SWVoiceIn) entries;
};
+struct audio_driver {
+ const char *name;
+ const char *descr;
+ struct audio_option *options;
+ void *(*init) (void);
+ void (*fini) (void *);
+ struct audio_pcm_ops *pcm_ops;
+ int can_be_default;
+ int max_voices_out;
+ int max_voices_in;
+ int voice_size_out;
+ int voice_size_in;
+};
+
struct audio_pcm_ops {
- int (*init_out)(HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt);
+ int (*init_out)(HWVoiceOut *hw, audsettings_t *as);
void (*fini_out)(HWVoiceOut *hw);
int (*run_out) (HWVoiceOut *hw);
int (*write) (SWVoiceOut *sw, void *buf, int size);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
- int (*init_in) (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt);
+ int (*init_in) (HWVoiceIn *hw, audsettings_t *as);
void (*fini_in) (HWVoiceIn *hw);
int (*run_in) (HWVoiceIn *hw);
int (*read) (SWVoiceIn *sw, void *buf, int size);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
-void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
- int nchannels, audfmt_e fmt, int swap_endian);
+struct AudioState {
+ struct audio_driver *drv;
+ void *drv_opaque;
+
+ QEMUTimer *ts;
+ LIST_HEAD (card_head, QEMUSoundCard) card_head;
+ LIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
+ LIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
+ int nb_hw_voices_out;
+ int nb_hw_voices_in;
+};
+
+extern struct audio_driver no_audio_driver;
+extern struct audio_driver oss_audio_driver;
+extern struct audio_driver sdl_audio_driver;
+extern struct audio_driver wav_audio_driver;
+extern struct audio_driver fmod_audio_driver;
+extern struct audio_driver alsa_audio_driver;
+extern struct audio_driver coreaudio_audio_driver;
+extern struct audio_driver dsound_audio_driver;
+extern volume_t nominal_volume;
+
+void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as,
+ int swap_endian);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
@@ -217,6 +193,9 @@
int audio_pcm_hw_get_live_out (HWVoiceOut *hw);
int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live);
+int audio_bug (const char *funcname, int cond);
+void *audio_calloc (const char *funcname, int nmemb, size_t size);
+
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 25ea72f..d985c2e 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -32,6 +32,43 @@
#define SW glue (SWVoice, In)
#endif
+static int glue (audio_pcm_hw_init_, TYPE) (
+ HW *hw,
+ audsettings_t *as
+ )
+{
+ glue (audio_pcm_hw_free_resources_, TYPE) (hw);
+
+ if (glue (hw->pcm_ops->init_, TYPE) (hw, as)) {
+ return -1;
+ }
+
+ if (audio_bug (AUDIO_FUNC, hw->samples <= 0)) {
+ dolog ("hw->samples=%d\n", hw->samples);
+ return -1;
+ }
+
+ LIST_INIT (&hw->sw_head);
+#ifdef DAC
+ hw->clip =
+ mixeng_clip
+#else
+ hw->conv =
+ mixeng_conv
+#endif
+ [hw->info.nchannels == 2]
+ [hw->info.sign]
+ [hw->info.swap_endian]
+ [hw->info.bits == 16];
+
+ if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
+ glue (hw->pcm_ops->fini_, TYPE) (hw);
+ return -1;
+ }
+
+ return 0;
+}
+
static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
{
glue (audio_pcm_sw_free_resources_, TYPE) (sw);
@@ -51,89 +88,86 @@
LIST_REMOVE (sw, entries);
}
-static void glue (audio_pcm_hw_fini_, TYPE) (HW *hw)
+static void glue (audio_pcm_hw_gc_, TYPE) (AudioState *s, HW **hwp)
{
- if (hw->active) {
+ HW *hw = *hwp;
+
+ if (!hw->sw_head.lh_first) {
+ LIST_REMOVE (hw, entries);
+ glue (s->nb_hw_voices_, TYPE) += 1;
glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
glue (hw->pcm_ops->fini_, TYPE) (hw);
- memset (hw, 0, glue (audio_state.drv->voice_size_, TYPE));
+ qemu_free (hw);
+ *hwp = NULL;
}
}
-static void glue (audio_pcm_hw_gc_, TYPE) (HW *hw)
+static HW *glue (audio_pcm_hw_find_any_, TYPE) (AudioState *s, HW *hw)
{
- if (!hw->sw_head.lh_first) {
- glue (audio_pcm_hw_fini_, TYPE) (hw);
- }
+ return hw ? hw->entries.le_next : s->glue (hw_head_, TYPE).lh_first;
}
-static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (AudioState *s, HW *hw)
{
- return hw ? hw->entries.le_next : glue (hw_head_, TYPE).lh_first;
-}
-
-static HW *glue (audio_pcm_hw_find_any_active_, TYPE) (HW *hw)
-{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
- if (hw->active) {
+ while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (s, hw))) {
+ if (hw->enabled) {
return hw;
}
}
return NULL;
}
-static HW *glue (audio_pcm_hw_find_any_active_enabled_, TYPE) (HW *hw)
+static HW *glue (audio_pcm_hw_find_any_passive_, TYPE) (AudioState *s)
{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
- if (hw->active && hw->enabled) {
- return hw;
- }
- }
- return NULL;
-}
+ if (glue (s->nb_hw_voices_, TYPE)) {
+ struct audio_driver *drv = s->drv;
-static HW *glue (audio_pcm_hw_find_any_passive_, TYPE) (HW *hw)
-{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
- if (!hw->active) {
- return hw;
+ if (audio_bug (AUDIO_FUNC, !drv)) {
+ dolog ("No host audio driver\n");
+ return NULL;
}
+
+ HW *hw = audio_calloc (AUDIO_FUNC, 1, glue (drv->voice_size_, TYPE));
+ if (!hw) {
+ dolog ("Can not allocate voice `%s' size %d\n",
+ drv->name, glue (drv->voice_size_, TYPE));
+ }
+
+ LIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
+ glue (s->nb_hw_voices_, TYPE) -= 1;
+ return hw;
}
+
return NULL;
}
static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
+ AudioState *s,
HW *hw,
- int freq,
- int nchannels,
- audfmt_e fmt
+ audsettings_t *as
)
{
- while ((hw = glue (audio_pcm_hw_find_any_active_, TYPE) (hw))) {
- if (audio_pcm_info_eq (&hw->info, freq, nchannels, fmt)) {
+ while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (s, hw))) {
+ if (audio_pcm_info_eq (&hw->info, as)) {
return hw;
}
}
return NULL;
}
-static HW *glue (audio_pcm_hw_add_new_, TYPE) (
- int freq,
- int nchannels,
- audfmt_e fmt
- )
+static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
{
HW *hw;
- hw = glue (audio_pcm_hw_find_any_passive_, TYPE) (NULL);
+ hw = glue (audio_pcm_hw_find_any_passive_, TYPE) (s);
if (hw) {
- hw->pcm_ops = audio_state.drv->pcm_ops;
+ hw->pcm_ops = s->drv->pcm_ops;
if (!hw->pcm_ops) {
return NULL;
}
- if (glue (audio_pcm_hw_init_, TYPE) (hw, freq, nchannels, fmt)) {
- glue (audio_pcm_hw_gc_, TYPE) (hw);
+ if (glue (audio_pcm_hw_init_, TYPE) (hw, as)) {
+ glue (audio_pcm_hw_gc_, TYPE) (s, &hw);
return NULL;
}
else {
@@ -144,66 +178,62 @@
return NULL;
}
-static HW *glue (audio_pcm_hw_add_, TYPE) (
- int freq,
- int nchannels,
- audfmt_e fmt
- )
+static HW *glue (audio_pcm_hw_add_, TYPE) (AudioState *s, audsettings_t *as)
{
HW *hw;
- if (glue (audio_state.greedy_, TYPE)) {
- hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+ if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) {
+ hw = glue (audio_pcm_hw_add_new_, TYPE) (s, as);
if (hw) {
return hw;
}
}
- hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, freq, nchannels, fmt);
+ hw = glue (audio_pcm_hw_find_specific_, TYPE) (s, NULL, as);
if (hw) {
return hw;
}
- hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+ hw = glue (audio_pcm_hw_add_new_, TYPE) (s, as);
if (hw) {
return hw;
}
- return glue (audio_pcm_hw_find_any_active_, TYPE) (NULL);
+ return glue (audio_pcm_hw_find_any_, TYPE) (s, NULL);
}
static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
- const char *name,
- int freq,
- int nchannels,
- audfmt_e fmt
+ AudioState *s,
+ const char *sw_name,
+ audsettings_t *as
)
{
SW *sw;
HW *hw;
- int hw_freq = freq;
- int hw_nchannels = nchannels;
- int hw_fmt = fmt;
+ audsettings_t hw_as;
- if (glue (audio_state.fixed_settings_, TYPE)) {
- hw_freq = glue (audio_state.fixed_freq_, TYPE);
- hw_nchannels = glue (audio_state.fixed_channels_, TYPE);
- hw_fmt = glue (audio_state.fixed_fmt_, TYPE);
+ if (glue (conf.fixed_, TYPE).enabled) {
+ hw_as = glue (conf.fixed_, TYPE).settings;
+ }
+ else {
+ hw_as = *as;
}
- sw = qemu_mallocz (sizeof (*sw));
+ sw = audio_calloc (AUDIO_FUNC, 1, sizeof (*sw));
if (!sw) {
+ dolog ("Could not allocate soft voice `%s' (%d bytes)\n",
+ sw_name ? sw_name : "unknown", sizeof (*sw));
goto err1;
}
- hw = glue (audio_pcm_hw_add_, TYPE) (hw_freq, hw_nchannels, hw_fmt);
+ hw = glue (audio_pcm_hw_add_, TYPE) (s, &hw_as);
if (!hw) {
goto err2;
}
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
- if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, freq, nchannels, fmt)) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
goto err3;
}
@@ -211,67 +241,86 @@
err3:
glue (audio_pcm_hw_del_sw_, TYPE) (sw);
- glue (audio_pcm_hw_gc_, TYPE) (hw);
+ glue (audio_pcm_hw_gc_, TYPE) (s, &hw);
err2:
qemu_free (sw);
err1:
return NULL;
}
-void glue (AUD_close_, TYPE) (SW *sw)
+static void glue (audio_close_, TYPE) (AudioState *s, SW *sw)
+{
+ glue (audio_pcm_sw_fini_, TYPE) (sw);
+ glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+ glue (audio_pcm_hw_gc_, TYPE) (s, &sw->hw);
+ qemu_free (sw);
+}
+void glue (AUD_close_, TYPE) (QEMUSoundCard *card, SW *sw)
{
if (sw) {
- glue (audio_pcm_sw_fini_, TYPE) (sw);
- glue (audio_pcm_hw_del_sw_, TYPE) (sw);
- glue (audio_pcm_hw_gc_, TYPE) (sw->hw);
- qemu_free (sw);
+ if (audio_bug (AUDIO_FUNC, !card || !card->audio)) {
+ dolog ("card=%p card->audio=%p\n",
+ card, card ? card->audio : NULL);
+ return;
+ }
+
+ glue (audio_close_, TYPE) (card->audio, sw);
}
}
SW *glue (AUD_open_, TYPE) (
+ QEMUSoundCard *card,
SW *sw,
const char *name,
void *callback_opaque ,
audio_callback_fn_t callback_fn,
- int freq,
- int nchannels,
- audfmt_e fmt
+ audsettings_t *as
)
{
+ AudioState *s;
#ifdef DAC
int live = 0;
SW *old_sw = NULL;
#endif
- if (!callback_fn) {
- dolog ("No callback specifed for voice `%s'\n", name);
+ ldebug ("open %s, freq %d, nchannels %d, fmt %d\n",
+ name, as->freq, as->nchannels, as->fmt);
+
+ if (audio_bug (AUDIO_FUNC,
+ !card || !card->audio || !name || !callback_fn || !as)) {
+ dolog ("card=%p card->audio=%p name=%p callback_fn=%p as=%p\n",
+ card, card ? card->audio : NULL, name, callback_fn, as);
goto fail;
}
- if (nchannels != 1 && nchannels != 2) {
- dolog ("Bogus channel count %d for voice `%s'\n", nchannels, name);
+ s = card->audio;
+
+ if (audio_bug (AUDIO_FUNC, audio_validate_settigs (as))) {
+ audio_print_settings (as);
goto fail;
}
- if (!audio_state.drv) {
- dolog ("No audio driver defined\n");
+ if (audio_bug (AUDIO_FUNC, !s->drv)) {
+ dolog ("Can not open `%s' (no host audio driver)\n", name);
goto fail;
}
- if (sw && audio_pcm_info_eq (&sw->info, freq, nchannels, fmt)) {
+ if (sw && audio_pcm_info_eq (&sw->info, as)) {
return sw;
}
#ifdef DAC
- if (audio_state.plive && sw && (!sw->active && !sw->empty)) {
+ if (conf.plive && sw && (!sw->active && !sw->empty)) {
live = sw->total_hw_samples_mixed;
#ifdef DEBUG_PLIVE
- dolog ("Replacing voice %s with %d live samples\n", sw->name, live);
+ dolog ("Replacing voice %s with %d live samples\n", SW_NAME (sw), live);
dolog ("Old %s freq %d, bits %d, channels %d\n",
- sw->name, sw->info.freq, sw->info.bits, sw->info.nchannels);
+ SW_NAME (sw), sw->info.freq, sw->info.bits, sw->info.nchannels);
dolog ("New %s freq %d, bits %d, channels %d\n",
- name, freq, (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) ? 16 : 8,
+ name,
+ freq,
+ (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) ? 16 : 8,
nchannels);
#endif
@@ -283,8 +332,8 @@
}
#endif
- if (!glue (audio_state.fixed_settings_, TYPE) && sw) {
- glue (AUD_close_, TYPE) (sw);
+ if (!glue (conf.fixed_, TYPE).enabled && sw) {
+ glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
@@ -292,30 +341,19 @@
HW *hw = sw->hw;
if (!hw) {
- dolog ("Internal logic error voice %s has no hardware store\n",
- name);
+ dolog ("Internal logic error voice `%s' has no hardware store\n",
+ SW_NAME (sw));
goto fail;
}
- if (glue (audio_pcm_sw_init_, TYPE) (
- sw,
- hw,
- name,
- freq,
- nchannels,
- fmt
- )) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
goto fail;
}
}
else {
- sw = glue (audio_pcm_create_voice_pair_, TYPE) (
- name,
- freq,
- nchannels,
- fmt);
+ sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as);
if (!sw) {
- dolog ("Failed to create voice %s\n", name);
+ dolog ("Failed to create voice `%s'\n", name);
goto fail;
}
}
@@ -349,7 +387,7 @@
return sw;
fail:
- glue (AUD_close_, TYPE) (sw);
+ glue (AUD_close_, TYPE) (card, sw);
return NULL;
}
@@ -367,10 +405,7 @@
ts->old_ts = sw->hw->ts_helper;
}
-uint64_t glue (AUD_time_stamp_get_elapsed_usec_, TYPE) (
- SW *sw,
- QEMUAudioTimeStamp *ts
- )
+uint64_t glue (AUD_get_elapsed_usec_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
{
uint64_t delta, cur_ts, old_ts;
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index eee1238..8551938 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -31,8 +31,6 @@
#define AUDIO_CAP "coreaudio"
#include "audio_int.h"
-#define DEVICE_BUFFER_FRAMES (512)
-
struct {
int buffer_frames;
} conf = {
@@ -132,7 +130,7 @@
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
@@ -147,7 +145,7 @@
err = pthread_mutex_lock (&core->mutex);
if (err) {
- dolog ("Can not lock voice for %s\nReason: %s\n",
+ dolog ("Could not lock voice for %s\nReason: %s\n",
fn_name, strerror (err));
return -1;
}
@@ -160,7 +158,7 @@
err = pthread_mutex_unlock (&core->mutex);
if (err) {
- dolog ("Can not unlock voice for %s\nReason: %s\n",
+ dolog ("Could not unlock voice for %s\nReason: %s\n",
fn_name, strerror (err));
return -1;
}
@@ -268,8 +266,7 @@
return audio_pcm_sw_write (sw, buf, len);
}
-static int coreaudio_init_out (HWVoiceOut *hw, int freq,
- int nchannels, audfmt_e fmt)
+static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
@@ -282,25 +279,22 @@
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
if (err) {
- dolog("Can not create mutex\nReason: %s\n", strerror (err));
+ dolog("Could not create mutex\nReason: %s\n", strerror (err));
return -1;
}
- if (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) {
+ if (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) {
bits = 16;
endianess = 1;
}
audio_pcm_init_info (
&hw->info,
- freq,
- nchannels,
- fmt,
+ as,
/* Following is irrelevant actually since we do not use
mixengs clipping routines */
audio_need_to_swap_endian (endianess)
);
- hw->bufsize = 4 * conf.buffer_frames * nchannels * bits;
/* open default output device */
propertySize = sizeof(core->outputDeviceID);
@@ -310,18 +304,18 @@
&core->outputDeviceID);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
- "Can not get default output Device\n");
+ "Could not get default output Device\n");
return -1;
}
if (core->outputDeviceID == kAudioDeviceUnknown) {
- dolog ("Can not initialize %s - Unknown Audiodevice\n", typ);
+ dolog ("Could not initialize %s - Unknown Audiodevice\n", typ);
return -1;
}
/* set Buffersize to conf.buffer_frames frames */
propertySize = sizeof(core->audioDevicePropertyBufferSize);
core->audioDevicePropertyBufferSize =
- conf.buffer_frames * sizeof(float) * 2;
+ conf.buffer_frames * sizeof(float) << (as->nchannels == 2);
status = AudioDeviceSetProperty(
core->outputDeviceID,
NULL,
@@ -332,7 +326,7 @@
&core->audioDevicePropertyBufferSize);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
- "Can not set device buffer size %d\n",
+ "Could not set device buffer size %d\n",
kAudioDevicePropertyBufferSize);
return -1;
}
@@ -347,9 +341,11 @@
&propertySize,
&core->audioDevicePropertyBufferSize);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr2 (status, typ, "Can not get device buffer size\n");
+ coreaudio_logerr2 (status, typ, "Could not get device buffer size\n");
return -1;
}
+ hw->samples = (core->audioDevicePropertyBufferSize / sizeof (float))
+ >> (as->nchannels == 2);
/* get StreamFormat */
propertySize = sizeof(core->outputStreamBasicDescription);
@@ -362,13 +358,13 @@
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
- "Can not get Device Stream properties\n");
+ "Could not get Device Stream properties\n");
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
/* set Samplerate */
- core->outputStreamBasicDescription.mSampleRate = (Float64)freq;
+ core->outputStreamBasicDescription.mSampleRate = (Float64)as->freq;
propertySize = sizeof(core->outputStreamBasicDescription);
status = AudioDeviceSetProperty(
core->outputDeviceID,
@@ -379,7 +375,7 @@
propertySize,
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr2 (status, typ, "Can not set samplerate %d\n", freq);
+ coreaudio_logerr2 (status, typ, "Could not set samplerate %d\n", freq);
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
@@ -387,7 +383,7 @@
/* set Callback */
status = AudioDeviceAddIOProc(core->outputDeviceID, audioDeviceIOProc, hw);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr2 (status, typ, "Can not set IOProc\n");
+ coreaudio_logerr2 (status, typ, "Could not set IOProc\n");
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
@@ -396,7 +392,7 @@
if (!core->isPlaying) {
status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr2 (status, typ, "Can not start playback\n");
+ coreaudio_logerr2 (status, typ, "Could not start playback\n");
AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
@@ -417,7 +413,7 @@
if (core->isPlaying) {
status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr (status, "Can not stop playback\n");
+ coreaudio_logerr (status, "Could not stop playback\n");
}
core->isPlaying = 0;
}
@@ -425,14 +421,14 @@
/* remove callback */
status = AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr (status, "Can not remove IOProc\n");
+ coreaudio_logerr (status, "Could not remove IOProc\n");
}
core->outputDeviceID = kAudioDeviceUnknown;
/* destroy mutex */
err = pthread_mutex_destroy(&core->mutex);
if (err) {
- dolog("Can not destroy mutex\nReason: %s\n", strerror (err));
+ dolog("Could not destroy mutex\nReason: %s\n", strerror (err));
}
}
@@ -447,7 +443,7 @@
if (!core->isPlaying) {
status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr (status, "Can not unpause playback\n");
+ coreaudio_logerr (status, "Could not unpause playback\n");
}
core->isPlaying = 1;
}
@@ -458,7 +454,7 @@
if (core->isPlaying) {
status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
- coreaudio_logerr (status, "Can not pause playback\n");
+ coreaudio_logerr (status, "Could not pause playback\n");
}
core->isPlaying = 0;
}
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index a04806e..38ba5b9 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -47,7 +47,7 @@
hr = glue (IFACE, _Unlock) (buf, p1, blen1, p2, blen2);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not unlock " NAME "\n");
+ dsound_logerr (hr, "Could not unlock " NAME "\n");
return -1;
}
@@ -93,13 +93,13 @@
#ifndef DSBTYPE_IN
if (hr == DSERR_BUFFERLOST) {
if (glue (dsound_restore_, TYPE) (buf)) {
- dsound_logerr (hr, "Can not lock " NAME "\n");
+ dsound_logerr (hr, "Could not lock " NAME "\n");
goto fail;
}
continue;
}
#endif
- dsound_logerr (hr, "Can not lock " NAME "\n");
+ dsound_logerr (hr, "Could not lock " NAME "\n");
goto fail;
}
@@ -158,38 +158,28 @@
if (ds->FIELD) {
hr = glue (IFACE, _Stop) (ds->FIELD);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not stop " NAME "\n");
+ dsound_logerr (hr, "Could not stop " NAME "\n");
}
hr = glue (IFACE, _Release) (ds->FIELD);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not release " NAME "\n");
+ dsound_logerr (hr, "Could not release " NAME "\n");
}
ds->FIELD = NULL;
}
}
#ifdef DSBTYPE_IN
-static int dsound_init_in (
- HWVoiceIn *hw,
- int freq,
- int nchannels,
- audfmt_e fmt
- )
+static int dsound_init_in (HWVoiceIn *hw, audsettings_t *as)
#else
-static int dsound_init_out (
- HWVoiceOut *hw,
- int freq,
- int nchannels,
- audfmt_e fmt
- )
+static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as)
#endif
{
int err;
HRESULT hr;
dsound *s = &glob_dsound;
WAVEFORMATEX wfx;
- struct full_fmt full_fmt;
+ audsettings_t obt_as;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
@@ -202,10 +192,7 @@
DSBCAPS bc;
#endif
- full_fmt.freq = freq;
- full_fmt.nchannels = nchannels;
- full_fmt.fmt = fmt;
- err = waveformat_from_full_fmt (&wfx, &full_fmt);
+ err = waveformat_from_audio_settings (&wfx, as);
if (err) {
return -1;
}
@@ -233,18 +220,13 @@
#endif
if (FAILED (hr)) {
- dsound_logerr2 (hr, typ, "Can not create " NAME "\n");
+ dsound_logerr2 (hr, typ, "Could not create " NAME "\n");
return -1;
}
- hr = glue (IFACE, _GetFormat) (
- ds->FIELD,
- &wfx,
- sizeof (wfx),
- NULL
- );
+ hr = glue (IFACE, _GetFormat) (ds->FIELD, &wfx, sizeof (wfx), NULL);
if (FAILED (hr)) {
- dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+ dsound_logerr2 (hr, typ, "Could not get " NAME " format\n");
goto fail0;
}
@@ -258,31 +240,33 @@
hr = glue (IFACE, _GetCaps) (ds->FIELD, &bc);
if (FAILED (hr)) {
- dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+ dsound_logerr2 (hr, typ, "Could not get " NAME " format\n");
goto fail0;
}
- err = waveformat_to_full_fmt (&wfx, &full_fmt);
+ err = waveformat_to_audio_settings (&wfx, &obt_as);
if (err) {
goto fail0;
}
ds->first_time = 1;
- hw->bufsize = bc.dwBufferBytes;
- audio_pcm_init_info (
- &hw->info,
- full_fmt.freq,
- full_fmt.nchannels,
- full_fmt.fmt,
- audio_need_to_swap_endian (0)
- );
+
+ audio_pcm_init_info (&hw->info, &obt_as, audio_need_to_swap_endian (0));
+
+ if (bc.dwBufferBytes & hw->info.align) {
+ dolog (
+ "GetCaps returned misaligned buffer size %ld, alignment %d\n",
+ bc.dwBufferBytes, hw->info.align + 1
+ );
+ }
+ hw->samples = bc.dwBufferBytes >> hw->info.shift;
#ifdef DEBUG_DSOUND
dolog ("caps %ld, desc %ld\n",
bc.dwBufferBytes, bd.dwBufferBytes);
dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
- hw->bufsize, full_fmt.freq, full_fmt.nchannels, full_fmt.fmt);
+ hw->bufsize, settings.freq, settings.nchannels, settings.fmt);
#endif
return 0;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 64b8417..63c5a50 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -37,12 +37,6 @@
/* #define DEBUG_DSOUND */
-struct full_fmt {
- int freq;
- int nchannels;
- audfmt_e fmt;
-};
-
static struct {
int lock_retries;
int restore_retries;
@@ -50,7 +44,7 @@
int set_primary;
int bufsize_in;
int bufsize_out;
- struct full_fmt full_fmt;
+ audsettings_t settings;
int latency_millis;
} conf = {
1,
@@ -71,7 +65,7 @@
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
LPDIRECTSOUNDBUFFER dsound_primary_buffer;
- struct full_fmt fmt;
+ audsettings_t settings;
} dsound;
static dsound glob_dsound;
@@ -259,7 +253,7 @@
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
@@ -301,7 +295,7 @@
continue;
default:
- dsound_logerr (hr, "Can not restore playback buffer\n");
+ dsound_logerr (hr, "Could not restore playback buffer\n");
return -1;
}
}
@@ -310,19 +304,18 @@
return -1;
}
-static int waveformat_from_full_fmt (WAVEFORMATEX *wfx,
- struct full_fmt *full_fmt)
+static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
{
memset (wfx, 0, sizeof (*wfx));
wfx->wFormatTag = WAVE_FORMAT_PCM;
- wfx->nChannels = full_fmt->nchannels;
- wfx->nSamplesPerSec = full_fmt->freq;
- wfx->nAvgBytesPerSec = full_fmt->freq << (full_fmt->nchannels == 2);
- wfx->nBlockAlign = 1 << (full_fmt->nchannels == 2);
+ wfx->nChannels = as->nchannels;
+ wfx->nSamplesPerSec = as->freq;
+ wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);
+ wfx->nBlockAlign = 1 << (as->nchannels == 2);
wfx->cbSize = 0;
- switch (full_fmt->fmt) {
+ switch (as->fmt) {
case AUD_FMT_S8:
wfx->wBitsPerSample = 8;
break;
@@ -344,16 +337,14 @@
break;
default:
- dolog ("Internal logic error: Bad audio format %d\n",
- full_fmt->freq);
+ dolog ("Internal logic error: Bad audio format %d\n", as->freq);
return -1;
}
return 0;
}
-static int waveformat_to_full_fmt (WAVEFORMATEX *wfx,
- struct full_fmt *full_fmt)
+static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
{
if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
dolog ("Invalid wave format, tag is not PCM, but %d\n",
@@ -365,15 +356,15 @@
dolog ("Invalid wave format, frequency is zero\n");
return -1;
}
- full_fmt->freq = wfx->nSamplesPerSec;
+ as->freq = wfx->nSamplesPerSec;
switch (wfx->nChannels) {
case 1:
- full_fmt->nchannels = 1;
+ as->nchannels = 1;
break;
case 2:
- full_fmt->nchannels = 2;
+ as->nchannels = 2;
break;
default:
@@ -386,11 +377,11 @@
switch (wfx->wBitsPerSample) {
case 8:
- full_fmt->fmt = AUD_FMT_U8;
+ as->fmt = AUD_FMT_U8;
break;
case 16:
- full_fmt->fmt = AUD_FMT_S16;
+ as->fmt = AUD_FMT_S16;
break;
default:
@@ -415,7 +406,7 @@
for (i = 0; i < conf.getstatus_retries; ++i) {
hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not get playback buffer status\n");
+ dsound_logerr (hr, "Could not get playback buffer status\n");
return -1;
}
@@ -438,7 +429,7 @@
hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not get capture buffer status\n");
+ dsound_logerr (hr, "Could not get capture buffer status\n");
return -1;
}
@@ -520,7 +511,7 @@
if (s->dsound_primary_buffer) {
hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not release primary buffer\n");
+ dsound_logerr (hr, "Could not release primary buffer\n");
}
s->dsound_primary_buffer = NULL;
}
@@ -542,7 +533,7 @@
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not set cooperative level for window %p\n",
+ dsound_logerr (hr, "Could not set cooperative level for window %p\n",
hwnd);
return -1;
}
@@ -551,7 +542,7 @@
return 0;
}
- err = waveformat_from_full_fmt (&wfx, &conf.full_fmt);
+ err = waveformat_from_audio_settings (&wfx, &conf.settings);
if (err) {
return -1;
}
@@ -569,13 +560,13 @@
NULL
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not create primary playback buffer\n");
+ dsound_logerr (hr, "Could not create primary playback buffer\n");
return -1;
}
hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not set primary playback buffer format\n");
+ dsound_logerr (hr, "Could not set primary playback buffer format\n");
}
hr = IDirectSoundBuffer_GetFormat (
@@ -585,7 +576,7 @@
NULL
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not get primary playback buffer format\n");
+ dsound_logerr (hr, "Could not get primary playback buffer format\n");
goto fail0;
}
@@ -594,7 +585,7 @@
print_wave_format (&wfx);
#endif
- err = waveformat_to_full_fmt (&wfx, &s->fmt);
+ err = waveformat_to_audio_settings (&wfx, &s->settings);
if (err) {
goto fail0;
}
@@ -625,7 +616,7 @@
}
if (status & DSBSTATUS_PLAYING) {
- dolog ("warning: voice is already playing\n");
+ dolog ("warning: Voice is already playing\n");
return 0;
}
@@ -633,7 +624,7 @@
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not start playing buffer\n");
+ dsound_logerr (hr, "Could not start playing buffer\n");
return -1;
}
break;
@@ -646,12 +637,12 @@
if (status & DSBSTATUS_PLAYING) {
hr = IDirectSoundBuffer_Stop (dsb);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not stop playing buffer\n");
+ dsound_logerr (hr, "Could not stop playing buffer\n");
return -1;
}
}
else {
- dolog ("warning: voice is not playing\n");
+ dolog ("warning: Voice is not playing\n");
}
break;
}
@@ -675,6 +666,7 @@
DWORD decr;
DWORD wpos, ppos, old_pos;
LPVOID p1, p2;
+ int bufsize;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -682,6 +674,7 @@
}
hwshift = hw->info.shift;
+ bufsize = hw->samples << hwshift;
live = audio_pcm_hw_get_live_out (hw);
@@ -691,7 +684,7 @@
ds->first_time ? &wpos : NULL
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not get playback buffer position\n");
+ dsound_logerr (hr, "Could not get playback buffer position\n");
return 0;
}
@@ -699,13 +692,14 @@
if (ds->first_time) {
if (conf.latency_millis) {
- DWORD cur_blat = audio_ring_dist (wpos, ppos, hw->bufsize);
+ DWORD cur_blat;
+ cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
- old_pos %= hw->bufsize;
+ old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
else {
@@ -734,14 +728,14 @@
len = ppos - old_pos;
}
else {
- if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->bufsize))) {
- len = hw->bufsize - old_pos + ppos;
+ if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
+ len = bufsize - old_pos + ppos;
}
}
- if (audio_bug (AUDIO_FUNC, len < 0 || len > hw->bufsize)) {
- dolog ("len=%d hw->bufsize=%d old_pos=%ld ppos=%ld\n",
- len, hw->bufsize, old_pos, ppos);
+ if (audio_bug (AUDIO_FUNC, len < 0 || len > bufsize)) {
+ dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
+ len, bufsize, old_pos, ppos);
return 0;
}
@@ -779,7 +773,7 @@
}
dsound_unlock_out (dsb, p1, p2, blen1, blen2);
- ds->old_pos = (old_pos + (decr << hwshift)) % hw->bufsize;
+ ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
#ifdef DEBUG_DSOUND
ds->mixed += decr << hwshift;
@@ -812,7 +806,7 @@
}
if (status & DSCBSTATUS_CAPTURING) {
- dolog ("warning: voice is already capturing\n");
+ dolog ("warning: Voice is already capturing\n");
return 0;
}
@@ -820,7 +814,7 @@
hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not start capturing\n");
+ dsound_logerr (hr, "Could not start capturing\n");
return -1;
}
break;
@@ -833,12 +827,12 @@
if (status & DSCBSTATUS_CAPTURING) {
hr = IDirectSoundCaptureBuffer_Stop (dscb);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not stop capturing\n");
+ dsound_logerr (hr, "Could not stop capturing\n");
return -1;
}
}
else {
- dolog ("warning: voice is not capturing\n");
+ dolog ("warning: Voice is not capturing\n");
}
break;
}
@@ -883,21 +877,21 @@
ds->first_time ? &rpos : NULL
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not get capture buffer position\n");
+ dsound_logerr (hr, "Could not get capture buffer position\n");
return 0;
}
if (ds->first_time) {
ds->first_time = 0;
if (rpos & hw->info.align) {
- ldebug ("warning: misaligned capture read position %ld(%d)\n",
+ ldebug ("warning: Misaligned capture read position %ld(%d)\n",
rpos, hw->info.align);
}
hw->wpos = rpos >> hwshift;
}
if (cpos & hw->info.align) {
- ldebug ("warning: misaligned capture position %ld(%d)\n",
+ ldebug ("warning: Misaligned capture position %ld(%d)\n",
cpos, hw->info.align);
}
cpos >>= hwshift;
@@ -951,7 +945,7 @@
hr = IDirectSound_Release (s->dsound);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not release DirectSound\n");
+ dsound_logerr (hr, "Could not release DirectSound\n");
}
s->dsound = NULL;
@@ -961,7 +955,7 @@
hr = IDirectSoundCapture_Release (s->dsound_capture);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not release DirectSoundCapture\n");
+ dsound_logerr (hr, "Could not release DirectSoundCapture\n");
}
s->dsound_capture = NULL;
}
@@ -974,7 +968,7 @@
hr = CoInitialize (NULL);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not initialize COM\n");
+ dsound_logerr (hr, "Could not initialize COM\n");
return NULL;
}
@@ -986,13 +980,13 @@
(void **) &s->dsound
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not create DirectSound instance\n");
+ dsound_logerr (hr, "Could not create DirectSound instance\n");
return NULL;
}
hr = IDirectSound_Initialize (s->dsound, NULL);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not initialize DirectSound\n");
+ dsound_logerr (hr, "Could not initialize DirectSound\n");
return NULL;
}
@@ -1004,16 +998,16 @@
(void **) &s->dsound_capture
);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not create DirectSoundCapture instance\n");
+ dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
}
else {
hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not initialize DirectSoundCapture\n");
+ dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");
hr = IDirectSoundCapture_Release (s->dsound_capture);
if (FAILED (hr)) {
- dsound_logerr (hr, "Can not release DirectSoundCapture\n");
+ dsound_logerr (hr, "Could not release DirectSoundCapture\n");
}
s->dsound_capture = NULL;
}
@@ -1039,11 +1033,11 @@
"Set the parameters of primary buffer", NULL, 0},
{"LATENCY_MILLIS", AUD_OPT_INT, &conf.latency_millis,
"(undocumented)", NULL, 0},
- {"PRIMARY_FREQ", AUD_OPT_INT, &conf.full_fmt.freq,
+ {"PRIMARY_FREQ", AUD_OPT_INT, &conf.settings.freq,
"Primary buffer frequency", NULL, 0},
- {"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.full_fmt.nchannels,
+ {"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.settings.nchannels,
"Primary buffer number of channels (1 - mono, 2 - stereo)", NULL, 0},
- {"PRIMARY_FMT", AUD_OPT_FMT, &conf.full_fmt.fmt,
+ {"PRIMARY_FMT", AUD_OPT_FMT, &conf.settings.fmt,
"Primary buffer format", NULL, 0},
{"BUFSIZE_OUT", AUD_OPT_INT, &conf.bufsize_out,
"(undocumented)", NULL, 0},
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 36b8d47..072d8a8 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -78,7 +78,7 @@
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
@@ -356,17 +356,17 @@
}
}
-static int fmod_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int fmod_init_out (HWVoiceOut *hw, audsettings_t *as)
{
int bits16, mode, channel;
FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
- mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+ mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
fmd->fmod_sample = FSOUND_Sample_Alloc (
FSOUND_FREE, /* index */
conf.nb_samples, /* length */
mode, /* mode */
- freq, /* freq */
+ as->freq, /* freq */
255, /* volume */
128, /* pan */
255 /* priority */
@@ -386,10 +386,9 @@
fmd->channel = channel;
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
- audio_need_to_swap_endian (0));
+ audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
bits16 = (mode & FSOUND_16BITS) != 0;
- hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+ hw->samples = conf.nb_samples;
return 0;
}
@@ -417,7 +416,7 @@
return 0;
}
-static int fmod_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+static int fmod_init_in (HWVoiceIn *hw, audsettings_t *as)
{
int bits16, mode;
FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
@@ -426,12 +425,12 @@
return -1;
}
- mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+ mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
fmd->fmod_sample = FSOUND_Sample_Alloc (
FSOUND_FREE, /* index */
conf.nb_samples, /* length */
mode, /* mode */
- freq, /* freq */
+ as->freq, /* freq */
255, /* volume */
128, /* pan */
255 /* priority */
@@ -443,10 +442,9 @@
}
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
- audio_need_to_swap_endian (0));
+ audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
bits16 = (mode & FSOUND_16BITS) != 0;
- hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+ hw->samples = conf.nb_samples;
return 0;
}
@@ -479,7 +477,7 @@
new_pos = FSOUND_Record_GetPosition ();
if (new_pos < 0) {
- fmod_logerr ("Can not get recording position\n");
+ fmod_logerr ("Could not get recording position\n");
return 0;
}
diff --git a/audio/mixeng.c b/audio/mixeng.c
index d43c5e5..14e37ae 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -228,21 +228,22 @@
*/
/* Private data */
-typedef struct ratestuff {
+struct rate {
uint64_t opos;
uint64_t opos_inc;
uint32_t ipos; /* position in the input stream (integer) */
st_sample_t ilast; /* last sample in the input stream */
-} *rate_t;
+};
/*
* Prepare processing.
*/
void *st_rate_start (int inrate, int outrate)
{
- rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
+ struct rate *rate = audio_calloc (AUDIO_FUNC, 1, sizeof (*rate));
if (!rate) {
+ dolog ("Could not allocate resampler (%d bytes)\n", sizeof (*rate));
return NULL;
}
diff --git a/audio/noaudio.c b/audio/noaudio.c
index e7936cc..aa35811 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -67,11 +67,10 @@
return audio_pcm_sw_write (sw, buf, len);
}
-static int no_init_out (HWVoiceOut *hw, int freq,
- int nchannels, audfmt_e fmt)
+static int no_init_out (HWVoiceOut *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
- hw->bufsize = 4096;
+ audio_pcm_init_info (&hw->info, as, 0);
+ hw->samples = 1024;
return 0;
}
@@ -87,11 +86,10 @@
return 0;
}
-static int no_init_in (HWVoiceIn *hw, int freq,
- int nchannels, audfmt_e fmt)
+static int no_init_in (HWVoiceIn *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
- hw->bufsize = 4096;
+ audio_pcm_init_info (&hw->info, as, 0);
+ hw->samples = 1024;
return 0;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index ff1a034..5072742 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -91,7 +91,7 @@
{
va_list ap;
- AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
@@ -179,7 +179,7 @@
return 0;
}
-#ifdef DEBUG_MISMATCHES
+#if defined DEBUG_MISMATCHES || defined DEBUG
static void oss_dump_info (struct oss_params *req, struct oss_params *obt)
{
dolog ("parameter | requested value | obtained value\n");
@@ -253,16 +253,16 @@
obt->fragsize = abinfo.fragsize;
*pfd = fd;
+#ifdef DEBUG_MISMATCHES
if ((req->fmt != obt->fmt) ||
(req->nchannels != obt->nchannels) ||
(req->freq != obt->freq) ||
(req->fragsize != obt->fragsize) ||
(req->nfrags != obt->nfrags)) {
-#ifdef DEBUG_MISMATCHES
dolog ("Audio parameters mismatch\n");
oss_dump_info (req, obt);
-#endif
}
+#endif
#ifdef DEBUG
oss_dump_info (req, obt);
@@ -283,12 +283,15 @@
st_sample_t *src;
struct audio_buf_info abinfo;
struct count_info cntinfo;
+ int bufsize;
live = audio_pcm_hw_get_live_out (hw);
if (!live) {
return 0;
}
+ bufsize = hw->samples << hw->info.shift;
+
if (oss->mmapped) {
int bytes;
@@ -300,7 +303,7 @@
if (cntinfo.ptr == oss->old_optr) {
if (abs (hw->samples - live) < 64) {
- dolog ("warning: overrun\n");
+ dolog ("warning: Overrun\n");
}
return 0;
}
@@ -309,7 +312,7 @@
bytes = cntinfo.ptr - oss->old_optr;
}
else {
- bytes = hw->bufsize + cntinfo.ptr - oss->old_optr;
+ bytes = bufsize + cntinfo.ptr - oss->old_optr;
}
decr = audio_MIN (bytes >> hw->info.shift, live);
@@ -321,9 +324,9 @@
return 0;
}
- if (abinfo.bytes < 0 || abinfo.bytes > hw->bufsize) {
- ldebug ("warning: invalid available size, size=%d bufsize=%d\n",
- abinfo.bytes, hw->bufsize);
+ if (abinfo.bytes < 0 || abinfo.bytes > bufsize) {
+ ldebug ("warning: Invalid available size, size=%d bufsize=%d\n",
+ abinfo.bytes, bufsize);
return 0;
}
@@ -362,7 +365,7 @@
int wsamples = written >> hw->info.shift;
int wbytes = wsamples << hw->info.shift;
if (wbytes != written) {
- dolog ("warning: misaligned write %d (requested %d), "
+ dolog ("warning: Misaligned write %d (requested %d), "
"alignment %d\n",
wbytes, written, hw->info.align + 1);
}
@@ -396,10 +399,10 @@
if (oss->pcm_buf) {
if (oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->bufsize);
+ err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
- oss->pcm_buf, hw->bufsize);
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
else {
@@ -409,7 +412,7 @@
}
}
-static int oss_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int oss_init_out (HWVoiceOut *hw, audsettings_t *as)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
struct oss_params req, obt;
@@ -417,10 +420,11 @@
int err;
int fd;
audfmt_e effective_fmt;
+ audsettings_t obt_as;
- req.fmt = aud_to_ossfmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_ossfmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.fragsize = conf.fragsize;
req.nfrags = conf.nfrags;
@@ -434,24 +438,38 @@
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- hw->bufsize = obt.nfrags * obt.fragsize;
+
+ if (obt.nfrags * obt.fragsize & hw->info.align) {
+ dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
+ obt.nfrags * obt.fragsize, hw->info.align + 1);
+ }
+
+ hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
if (conf.try_mmap) {
- oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE,
- MAP_SHARED, fd, 0);
+ oss->pcm_buf = mmap (
+ 0,
+ hw->samples << hw->info.shift,
+ PROT_READ | PROT_WRITE,
+ MAP_SHARED,
+ fd,
+ 0
+ );
if (oss->pcm_buf == MAP_FAILED) {
oss_logerr (errno, "Failed to map %d bytes of DAC\n",
- hw->bufsize);
+ hw->samples << hw->info.shift);
} else {
int err;
int trig = 0;
@@ -472,18 +490,24 @@
}
if (!oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->bufsize);
+ err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
- oss->pcm_buf, hw->bufsize);
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
}
}
if (!oss->mmapped) {
- oss->pcm_buf = qemu_mallocz (hw->bufsize);
+ oss->pcm_buf = audio_calloc (
+ AUDIO_FUNC,
+ hw->samples,
+ 1 << hw->info.shift
+ );
if (!oss->pcm_buf) {
+ dolog ("Could not allocate DAC buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
oss_anal_close (&fd);
return -1;
}
@@ -528,8 +552,7 @@
return 0;
}
-static int oss_init_in (HWVoiceIn *hw,
- int freq, int nchannels, audfmt_e fmt)
+static int oss_init_in (HWVoiceIn *hw, audsettings_t *as)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
struct oss_params req, obt;
@@ -537,10 +560,11 @@
int err;
int fd;
audfmt_e effective_fmt;
+ audsettings_t obt_as;
- req.fmt = aud_to_ossfmt (fmt);
- req.freq = freq;
- req.nchannels = nchannels;
+ req.fmt = aud_to_ossfmt (as->fmt);
+ req.freq = as->freq;
+ req.nchannels = as->nchannels;
req.fragsize = conf.fragsize;
req.nfrags = conf.nfrags;
if (oss_open (1, &req, &obt, &fd)) {
@@ -553,18 +577,28 @@
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.nchannels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.nchannels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianness)
);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- hw->bufsize = obt.nfrags * obt.fragsize;
- oss->pcm_buf = qemu_mallocz (hw->bufsize);
+
+ if (obt.nfrags * obt.fragsize & hw->info.align) {
+ dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
+ obt.nfrags * obt.fragsize, hw->info.align + 1);
+ }
+
+ hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!oss->pcm_buf) {
+ dolog ("Could not allocate ADC buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
oss_anal_close (&fd);
return -1;
}
@@ -623,7 +657,7 @@
if (nread > 0) {
if (nread & hw->info.align) {
- dolog ("warning: misaligned read %d (requested %d), "
+ dolog ("warning: Misaligned read %d (requested %d), "
"alignment %d\n", nread, bufs[i].add << hwshift,
hw->info.align + 1);
}
diff --git a/audio/rate_template.h b/audio/rate_template.h
index 5cc95c8..3e0e77c 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -30,7 +30,7 @@
void NAME (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
int *isamp, int *osamp)
{
- rate_t rate = (rate_t) opaque;
+ struct rate *rate = opaque;
st_sample_t *istart, *iend;
st_sample_t *ostart, *oend;
st_sample_t ilast, icur, out;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 673e2a1..713c784 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -303,7 +303,7 @@
sdl_close (&glob_sdl);
}
-static int sdl_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int sdl_init_out (HWVoiceOut *hw, audsettings_t *as)
{
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDLAudioState *s = &glob_sdl;
@@ -312,18 +312,14 @@
int endianess;
int err;
audfmt_e effective_fmt;
+ audsettings_t obt_as;
- if (nchannels != 2) {
- dolog ("Can not init DAC. Bogus channel count %d\n", nchannels);
- return -1;
- }
+ shift <<= as->nchannels == 2;
- req.freq = freq;
- req.format = aud_to_sdlfmt (fmt, &shift);
- req.channels = nchannels;
+ req.freq = as->freq;
+ req.format = aud_to_sdlfmt (as->fmt, &shift);
+ req.channels = as->nchannels;
req.samples = conf.nb_samples;
- shift <<= nchannels == 2;
-
req.callback = sdl_callback;
req.userdata = sdl;
@@ -337,14 +333,16 @@
return -1;
}
+ obt_as.freq = obt.freq;
+ obt_as.nchannels = obt.channels;
+ obt_as.fmt = effective_fmt;
+
audio_pcm_init_info (
&hw->info,
- obt.freq,
- obt.channels,
- effective_fmt,
+ &obt_as,
audio_need_to_swap_endian (endianess)
);
- hw->bufsize = obt.samples << shift;
+ hw->samples = obt.samples;
s->initialized = 1;
s->exit = 0;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index e9bd878..4cc9ca7 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -35,9 +35,15 @@
} WAVVoiceOut;
static struct {
+ audsettings_t settings;
const char *wav_path;
} conf = {
- .wav_path = "qemu.wav"
+ {
+ 44100,
+ 2,
+ AUD_FMT_S16
+ },
+ "qemu.wav"
};
static int wav_run_out (HWVoiceOut *hw)
@@ -101,22 +107,22 @@
}
}
-static int wav_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+static int wav_init_out (HWVoiceOut *hw, audsettings_t *as)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int bits16;
+ int bits16 = 0, stereo = 0;
uint8_t hdr[] = {
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
+ audsettings_t wav_as = conf.settings;
- freq = audio_state.fixed_freq_out;
- fmt = audio_state.fixed_fmt_out;
- nchannels = audio_state.fixed_channels_out;
+ (void) as;
- switch (fmt) {
+ stereo = wav_as.nchannels == 2;
+ switch (wav_as.fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
bits16 = 0;
@@ -126,32 +132,24 @@
case AUD_FMT_U16:
bits16 = 1;
break;
-
- default:
- dolog ("Internal logic error bad format %d\n", fmt);
- return -1;
}
hdr[34] = bits16 ? 0x10 : 0x08;
- audio_pcm_init_info (
- &hw->info,
- freq,
- nchannels,
- bits16 ? AUD_FMT_S16 : AUD_FMT_U8,
- audio_need_to_swap_endian (0)
- );
- hw->bufsize = 4096;
- wav->pcm_buf = qemu_mallocz (hw->bufsize);
+
+ audio_pcm_init_info (&hw->info, &wav_as, audio_need_to_swap_endian (0));
+
+ hw->samples = 1024;
+ wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!wav->pcm_buf) {
- dolog ("Can not initialize WAV buffer of %d bytes\n",
- hw->bufsize);
+ dolog ("Could not allocate buffer (%d bytes)\n",
+ hw->samples << hw->info.shift);
return -1;
}
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
- le_store (hdr + 28, hw->info.freq << (bits16 + (nchannels == 2)), 4);
- le_store (hdr + 32, 1 << (bits16 + (nchannels == 2)), 2);
+ le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
+ le_store (hdr + 32, 1 << (bits16 + stereo), 2);
wav->f = fopen (conf.wav_path, "wb");
if (!wav->f) {
@@ -175,7 +173,7 @@
uint32_t rifflen = (wav->total_samples << stereo) + 36;
uint32_t datalen = wav->total_samples << stereo;
- if (!wav->f || !hw->active) {
+ if (!wav->f) {
return;
}
@@ -214,6 +212,15 @@
}
struct audio_option wav_options[] = {
+ {"FREQUENCY", AUD_OPT_INT, &conf.settings.freq,
+ "Frequency", NULL, 0},
+
+ {"FORMAT", AUD_OPT_FMT, &conf.settings.fmt,
+ "Format", NULL, 0},
+
+ {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &conf.settings.nchannels,
+ "Number of channels (1 - mono, 2 - stereo)", NULL, 0},
+
{"PATH", AUD_OPT_STR, &conf.wav_path,
"Path to wave file", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
diff --git a/hw/adlib.c b/hw/adlib.c
index 70de4ff..fa2a03d 100644
--- a/hw/adlib.c
+++ b/hw/adlib.c
@@ -53,6 +53,7 @@
} conf = {0x220, 44100};
typedef struct {
+ QEMUSoundCard card;
int ticking[2];
int enabled;
int active;
@@ -70,7 +71,7 @@
#endif
} AdlibState;
-static AdlibState adlib;
+static AdlibState glob_adlib;
static void adlib_stop_opl_timer (AdlibState *s, size_t n)
{
@@ -90,7 +91,7 @@
if (s->ticking[i]) {
uint64_t delta;
- delta = AUD_time_stamp_get_elapsed_usec_out (s->voice, &s->ats);
+ delta = AUD_get_elapsed_usec_out (s->voice, &s->ats);
ldebug (
"delta = %f dexp = %f expired => %d\n",
delta / 1000000.0,
@@ -141,10 +142,11 @@
static void timer_handler (int c, double interval_Sec)
{
- AdlibState *s = &adlib;
+ AdlibState *s = &glob_adlib;
unsigned n = c & 1;
#ifdef DEBUG
double interval;
+ int64_t exp;
#endif
if (interval_Sec == 0.0) {
@@ -262,16 +264,23 @@
s->active = 0;
s->enabled = 0;
+ AUD_remove_card (&s->card);
}
-void Adlib_init (void)
+int Adlib_init (AudioState *audio)
{
- AdlibState *s = &adlib;
+ AdlibState *s = &glob_adlib;
+ audsettings_t as;
+
+ if (!audio) {
+ dolog ("No audio state\n");
+ return -1;
+ }
#ifdef HAS_YMF262
if (YMF262Init (1, 14318180, conf.freq)) {
dolog ("YMF262Init %d failed\n", conf.freq);
- return;
+ return -1;
}
else {
YMF262SetTimerHandler (0, timer_handler, 0);
@@ -281,7 +290,7 @@
s->opl = OPLCreate (OPL_TYPE_YM3812, 3579545, conf.freq);
if (!s->opl) {
dolog ("OPLCreate %d failed\n", conf.freq);
- return;
+ return -1;
}
else {
OPLSetTimerHandler (s->opl, timer_handler, 0);
@@ -289,18 +298,23 @@
}
#endif
+ as.freq = conf.freq;
+ as.nchannels = SHIFT;
+ as.fmt = AUD_FMT_S16;
+
+ AUD_register_card (audio, "adlib", &s->card);
+
s->voice = AUD_open_out (
+ &s->card,
s->voice,
"adlib",
s,
adlib_callback,
- conf.freq,
- SHIFT,
- AUD_FMT_S16
+ &as
);
if (!s->voice) {
Adlib_fini (s);
- return;
+ return -1;
}
s->samples = AUD_get_buffer_size_out (s->voice) >> SHIFT;
@@ -310,7 +324,7 @@
dolog ("not enough memory for adlib mixing buffer (%d)\n",
s->samples << SHIFT);
Adlib_fini (s);
- return;
+ return -1;
}
register_ioport_read (0x388, 4, 1, adlib_read, s);
@@ -321,4 +335,6 @@
register_ioport_read (conf.port + 8, 2, 1, adlib_read, s);
register_ioport_write (conf.port + 8, 2, 1, adlib_write, s);
+
+ return 0;
}
diff --git a/hw/es1370.c b/hw/es1370.c
index 0191f5f..fc7ac0a 100644
--- a/hw/es1370.c
+++ b/hw/es1370.c
@@ -265,6 +265,7 @@
typedef struct ES1370State {
PCIDevice *pci_dev;
+ QEMUSoundCard card;
struct chan chan[NB_CHANNELS];
SWVoiceOut *dac_voice[2];
SWVoiceIn *adc_voice;
@@ -341,11 +342,11 @@
d->scount = 0;
d->leftover = 0;
if (i == ADC_CHANNEL) {
- AUD_close_in (s->adc_voice);
+ AUD_close_in (&s->card, s->adc_voice);
s->adc_voice = NULL;
}
else {
- AUD_close_out (s->dac_voice[i]);
+ AUD_close_out (&s->card, s->dac_voice[i]);
s->dac_voice[i] = NULL;
}
}
@@ -417,28 +418,32 @@
(new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
d->shift);
if (new_freq) {
+ audsettings_t as;
+
+ as.freq = new_freq;
+ as.nchannels = 1 << (new_fmt & 1);
+ as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+
if (i == ADC_CHANNEL) {
s->adc_voice =
AUD_open_in (
+ &s->card,
s->adc_voice,
"es1370.adc",
s,
es1370_adc_callback,
- new_freq,
- 1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8
+ &as
);
}
else {
s->dac_voice[i] =
AUD_open_out (
+ &s->card,
s->dac_voice[i],
i ? "es1370.dac2" : "es1370.dac1",
s,
i ? es1370_dac2_callback : es1370_dac1_callback,
- new_freq,
- 1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8
+ &as
);
}
}
@@ -761,7 +766,7 @@
while (temp) {
int acquired, to_copy;
- to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
acquired = AUD_read (s->adc_voice, tmpbuf, to_copy);
if (!acquired)
break;
@@ -779,7 +784,7 @@
while (temp) {
int copied, to_copy;
- to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
cpu_physical_memory_read (addr, tmpbuf, to_copy);
copied = AUD_write (voice, tmpbuf, to_copy);
if (!copied)
@@ -812,7 +817,7 @@
else {
d->frame_cnt = size;
- if (cnt <= d->frame_cnt)
+ if ((uint32_t) cnt <= d->frame_cnt)
d->frame_cnt |= cnt << 16;
}
@@ -876,6 +881,10 @@
PCIES1370State *d = (PCIES1370State *) pci_dev;
ES1370State *s = &d->es1370;
+ (void) region_num;
+ (void) size;
+ (void) type;
+
register_ioport_write (addr, 0x40 * 4, 1, es1370_writeb, s);
register_ioport_write (addr, 0x40 * 2, 2, es1370_writew, s);
register_ioport_write (addr, 0x40, 4, es1370_writel, s);
@@ -923,13 +932,13 @@
qemu_get_be32s (f, &d->frame_cnt);
if (i == ADC_CHANNEL) {
if (s->adc_voice) {
- AUD_close_in (s->adc_voice);
+ AUD_close_in (&s->card, s->adc_voice);
s->adc_voice = NULL;
}
}
else {
if (s->dac_voice[i]) {
- AUD_close_out (s->dac_voice[i]);
+ AUD_close_out (&s->card, s->dac_voice[i]);
s->dac_voice[i] = NULL;
}
}
@@ -953,12 +962,22 @@
es1370_reset (s);
}
-int es1370_init (PCIBus *bus)
+int es1370_init (PCIBus *bus, AudioState *audio)
{
PCIES1370State *d;
ES1370State *s;
uint8_t *c;
+ if (!bus) {
+ dolog ("No PCI bus\n");
+ return -1;
+ }
+
+ if (!audio) {
+ dolog ("No audio state\n");
+ return -1;
+ }
+
d = (PCIES1370State *) pci_register_device (bus, "ES1370",
sizeof (PCIES1370State),
-1, NULL, NULL);
@@ -1002,6 +1021,8 @@
pci_register_io_region (&d->dev, 0, 256, PCI_ADDRESS_SPACE_IO, es1370_map);
register_savevm ("es1370", 0, 1, es1370_save, es1370_load, s);
qemu_register_reset (es1370_on_reset, s);
+
+ AUD_register_card (audio, "es1370", &s->card);
es1370_reset (s);
return 0;
}
diff --git a/hw/pc.c b/hw/pc.c
index 90a0e48..324f536 100644
--- a/hw/pc.c
+++ b/hw/pc.c
@@ -601,19 +601,23 @@
DMA_init(0);
if (audio_enabled) {
- AUD_init();
- if (sb16_enabled)
- SB16_init ();
+ AudioState *audio;
+
+ audio = AUD_init();
+ if (audio) {
+ if (sb16_enabled)
+ SB16_init (audio);
#ifdef CONFIG_ADLIB
- if (adlib_enabled)
- Adlib_init ();
+ if (adlib_enabled)
+ Adlib_init (audio);
#endif
#ifdef CONFIG_GUS
- if (gus_enabled)
- GUS_init ();
+ if (gus_enabled)
+ GUS_init (audio);
#endif
- if (pci_enabled && es1370_enabled)
- es1370_init (pci_bus);
+ if (pci_enabled && es1370_enabled)
+ es1370_init (pci_bus, audio);
+ }
}
floppy_controller = fdctrl_init(6, 2, 0, 0x3f0, fd_table);
diff --git a/hw/sb16.c b/hw/sb16.c
index e79d192..4414af3 100644
--- a/hw/sb16.c
+++ b/hw/sb16.c
@@ -53,6 +53,7 @@
} conf = {5, 4, 5, 1, 5, 0x220};
typedef struct SB16State {
+ QEMUSoundCard card;
int irq;
int dma;
int hdma;
@@ -108,9 +109,6 @@
uint8_t mixer_regs[256];
} SB16State;
-/* XXX: suppress that and use a context */
-static struct SB16State dsp;
-
static void SB_audio_callback (void *opaque, int free);
static int magic_of_irq (int irq)
@@ -242,15 +240,21 @@
s->block_size, s->dma_auto, s->fifo, s->highspeed);
if (s->freq) {
+ audsettings_t as;
+
s->audio_free = 0;
+
+ as.freq = s->freq;
+ as.nchannels = 1 << s->fmt_stereo;
+ as.fmt = s->fmt;
+
s->voice = AUD_open_out (
+ &s->card,
s->voice,
"sb16",
s,
SB_audio_callback,
- s->freq,
- 1 << s->fmt_stereo,
- s->fmt
+ &as
);
}
@@ -330,15 +334,21 @@
}
if (s->freq) {
+ audsettings_t as;
+
s->audio_free = 0;
+
+ as.freq = s->freq;
+ as.nchannels = 1 << s->fmt_stereo;
+ as.fmt = s->fmt;
+
s->voice = AUD_open_out (
+ &s->card,
s->voice,
"sb16",
s,
SB_audio_callback,
- s->freq,
- 1 << s->fmt_stereo,
- s->fmt
+ &as
);
}
@@ -349,7 +359,7 @@
static inline void dsp_out_data (SB16State *s, uint8_t val)
{
ldebug ("outdata %#x\n", val);
- if (s->out_data_len < sizeof (s->out_data)) {
+ if ((size_t) s->out_data_len < sizeof (s->out_data)) {
s->out_data[s->out_data_len++] = val;
}
}
@@ -1018,6 +1028,7 @@
static IO_WRITE_PROTO(mixer_write_indexb)
{
SB16State *s = opaque;
+ (void) nport;
s->mixer_nreg = val;
}
@@ -1025,10 +1036,8 @@
{
SB16State *s = opaque;
+ (void) nport;
ldebug ("mixer_write [%#x] <- %#x\n", s->mixer_nreg, val);
- if (s->mixer_nreg > sizeof (s->mixer_regs)) {
- return;
- }
switch (s->mixer_nreg) {
case 0x00:
@@ -1088,6 +1097,8 @@
static IO_READ_PROTO(mixer_read)
{
SB16State *s = opaque;
+
+ (void) nport;
#ifndef DEBUG_SB16_MOST
if (s->mixer_nreg != 0x82) {
ldebug ("mixer_read[%#x] -> %#x\n",
@@ -1111,11 +1122,12 @@
while (temp) {
int left = dma_len - dma_pos;
- int to_copy, copied;
+ int copied;
+ size_t to_copy;
to_copy = audio_MIN (temp, left);
- if (to_copy > sizeof(tmpbuf)) {
- to_copy = sizeof(tmpbuf);
+ if (to_copy > sizeof (tmpbuf)) {
+ to_copy = sizeof (tmpbuf);
}
copied = DMA_read_memory (nchan, tmpbuf, dma_pos, to_copy);
@@ -1308,21 +1320,27 @@
qemu_get_buffer (f, s->mixer_regs, 256);
if (s->voice) {
- AUD_close_out (s->voice);
+ AUD_close_out (&s->card, s->voice);
s->voice = NULL;
}
if (s->dma_running) {
if (s->freq) {
+ audsettings_t as;
+
s->audio_free = 0;
+
+ as.freq = s->freq;
+ as.nchannels = 1 << s->fmt_stereo;
+ as.fmt = s->fmt;
+
s->voice = AUD_open_out (
+ &s->card,
s->voice,
"sb16",
s,
SB_audio_callback,
- s->freq,
- 1 << s->fmt_stereo,
- s->fmt
+ &as
);
}
@@ -1332,13 +1350,25 @@
return 0;
}
-void SB16_init (void)
+int SB16_init (AudioState *audio)
{
- SB16State *s = &dsp;
+ SB16State *s;
int i;
static const uint8_t dsp_write_ports[] = {0x6, 0xc};
static const uint8_t dsp_read_ports[] = {0x6, 0xa, 0xc, 0xd, 0xe, 0xf};
+ if (!audio) {
+ dolog ("No audio state\n");
+ return -1;
+ }
+
+ s = qemu_mallocz (sizeof (*s));
+ if (!s) {
+ dolog ("Could not allocate memory for SB16 (%d bytes)\n",
+ sizeof (*s));
+ return -1;
+ }
+
s->cmd = -1;
s->irq = conf.irq;
s->dma = conf.dma;
@@ -1356,7 +1386,7 @@
reset_mixer (s);
s->aux_ts = qemu_new_timer (vm_clock, aux_timer, s);
if (!s->aux_ts) {
- dolog ("Can not create auxiliary timer\n");
+ dolog ("warning: Could not create auxiliary timer\n");
}
for (i = 0; i < LENOFA (dsp_write_ports); i++) {
@@ -1377,4 +1407,6 @@
s->can_write = 1;
register_savevm ("sb16", 0, 1, SB_save, SB_load, s);
+ AUD_register_card (audio, "sb16", &s->card);
+ return 0;
}
diff --git a/qemu-doc.texi b/qemu-doc.texi
index 51875d7..1c5b5b7 100644
--- a/qemu-doc.texi
+++ b/qemu-doc.texi
@@ -95,12 +95,21 @@
@item
Serial ports
@item
-Soundblaster 16 card
+Creative SoundBlaster 16 sound card
+@item
+ENSONIQ AudioPCI ES1370 sound card
+@item
+Adlib(OPL2) - Yamaha YM3812 compatible chip
@end itemize
+Note that adlib is only available when QEMU was configured with
+-enable-adlib
+
QEMU uses the PC BIOS from the Bochs project and the Plex86/Bochs LGPL
VGA BIOS.
+QEMU uses YM3812 emulation by Tatsuyuki Satoh.
+
@c man end
@section Quick Start
diff --git a/vl.c b/vl.c
index d551a4e..810db81 100644
--- a/vl.c
+++ b/vl.c
@@ -2842,10 +2842,11 @@
"-k language use keyboard layout (for example \"fr\" for French)\n"
#endif
#ifdef HAS_AUDIO
- "-enable-audio enable audio support\n"
+ "-enable-audio enable audio support, and all the sound cars\n"
"-audio-help print list of audio drivers and their options\n"
- "-soundhw c1,... comma separated list of sound card names\n"
- " use -soundhw ? to get the list of supported sound cards\n"
+ "-soundhw c1,... enable audio support\n"
+ " and only specified sound cards (comma separated list)\n"
+ " use -soundhw ? to get the list of supported cards\n"
#endif
"-localtime set the real time clock to local time [default=utc]\n"
"-full-screen start in full screen\n"
@@ -3145,9 +3146,9 @@
printf ("sb16 Creative Sound Blaster 16\n");
#ifdef CONFIG_ADLIB
#ifdef HAS_YMF262
- printf ("adlib Ymaha YMF262 (OPL3)\n");
+ printf ("adlib Yamaha YMF262 (OPL3)\n");
#else
- printf ("adlib Ymaha YM3812 (OPL2)\n");
+ printf ("adlib Yamaha YM3812 (OPL2)\n");
#endif
#endif
#ifdef CONFIG_GUS
diff --git a/vl.h b/vl.h
index e1a2ec3..72d4f87 100644
--- a/vl.h
+++ b/vl.h
@@ -631,16 +631,16 @@
SetIRQFunc *set_irq, void *irq_opaque, int irq);
/* es1370.c */
-int es1370_init (PCIBus *bus);
+int es1370_init (PCIBus *bus, AudioState *s);
/* sb16.c */
-void SB16_init (void);
+int SB16_init (AudioState *s);
/* adlib.c */
-void Adlib_init (void);
+int Adlib_init (AudioState *s);
/* gus.c */
-void GUS_init (void);
+int GUS_init (AudioState *s);
/* dma.c */
typedef int (*DMA_transfer_handler) (void *opaque, int nchan, int pos, int size);
