--- /dev/null
+/*
+ * QEMU Audio subsystem
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include "hw/hw.h"
+#include "audio.h"
+#include "monitor/monitor.h"
+#include "qemu/timer.h"
+#include "sysemu/sysemu.h"
+
+#define AUDIO_CAP "audio"
+#include "audio_int.h"
+
+/* #define DEBUG_LIVE */
+/* #define DEBUG_OUT */
+/* #define DEBUG_CAPTURE */
+/* #define DEBUG_POLL */
+
+#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
+
+
+/* Order of CONFIG_AUDIO_DRIVERS is import.
+ The 1st one is the one used by default, that is the reason
+ that we generate the list.
+*/
+static struct audio_driver *drvtab[] = {
+#ifdef CONFIG_SPICE
+ &spice_audio_driver,
+#endif
+ CONFIG_AUDIO_DRIVERS
+ &no_audio_driver,
+ &wav_audio_driver
+};
+
+struct fixed_settings {
+ int enabled;
+ int nb_voices;
+ int greedy;
+ struct audsettings settings;
+};
+
+static struct {
+ struct fixed_settings fixed_out;
+ struct fixed_settings fixed_in;
+ union {
+ int hertz;
+ int64_t ticks;
+ } period;
+ int try_poll_in;
+ int try_poll_out;
+} conf = {
+ .fixed_out = { /* DAC fixed settings */
+ .enabled = 1,
+ .nb_voices = 1,
+ .greedy = 1,
+ .settings = {
+ .freq = 44100,
+ .nchannels = 2,
+ .fmt = AUD_FMT_S16,
+ .endianness = AUDIO_HOST_ENDIANNESS,
+ }
+ },
+
+ .fixed_in = { /* ADC fixed settings */
+ .enabled = 1,
+ .nb_voices = 1,
+ .greedy = 1,
+ .settings = {
+ .freq = 44100,
+ .nchannels = 2,
+ .fmt = AUD_FMT_S16,
+ .endianness = AUDIO_HOST_ENDIANNESS,
+ }
+ },
+
+ .period = { .hertz = 100 },
+ .try_poll_in = 1,
+ .try_poll_out = 1,
+};
+
+static AudioState glob_audio_state;
+
+const struct mixeng_volume nominal_volume = {
+ .mute = 0,
+#ifdef FLOAT_MIXENG
+ .r = 1.0,
+ .l = 1.0,
+#else
+ .r = 1ULL << 32,
+ .l = 1ULL << 32,
+#endif
+};
+
+#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
+#error No its not
+#else
+static void audio_print_options (const char *prefix,
+ struct audio_option *opt);
+
+int audio_bug (const char *funcname, int cond)
+{
+ if (cond) {
+ static int shown;
+
+ AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
+ if (!shown) {
+ struct audio_driver *d;
+
+ shown = 1;
+ AUD_log (NULL, "Save all your work and restart without audio\n");
+ AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n");
+ AUD_log (NULL, "I am sorry\n");
+ d = glob_audio_state.drv;
+ if (d) {
+ audio_print_options (d->name, d->options);
+ }
+ }
+ AUD_log (NULL, "Context:\n");
+
+#if defined AUDIO_BREAKPOINT_ON_BUG
+# if defined HOST_I386
+# if defined __GNUC__
+ __asm__ ("int3");
+# elif defined _MSC_VER
+ _asm _emit 0xcc;
+# else
+ abort ();
+# endif
+# else
+ abort ();
+# endif
+#endif
+ }
+
+ return cond;
+}
+#endif
+
+static inline int audio_bits_to_index (int bits)
+{
+ switch (bits) {
+ case 8:
+ return 0;
+
+ case 16:
+ return 1;
+
+ case 32:
+ return 2;
+
+ default:
+ audio_bug ("bits_to_index", 1);
+ AUD_log (NULL, "invalid bits %d\n", bits);
+ return 0;
+ }
+}
+
+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=%zu (len=%zu)\n", nmemb, size, len);
+ return NULL;
+ }
+
+ return g_malloc0 (len);
+}
+
+static char *audio_alloc_prefix (const char *s)
+{
+ const char qemu_prefix[] = "QEMU_";
+ size_t len, i;
+ char *r, *u;
+
+ if (!s) {
+ return NULL;
+ }
+
+ len = strlen (s);
+ r = g_malloc (len + sizeof (qemu_prefix));
+
+ u = r + sizeof (qemu_prefix) - 1;
+
+ pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
+ pstrcat (r, len + sizeof (qemu_prefix), s);
+
+ for (i = 0; i < len; ++i) {
+ u[i] = qemu_toupper(u[i]);
+ }
+
+ return r;
+}
+
+static const char *audio_audfmt_to_string (audfmt_e fmt)
+{
+ switch (fmt) {
+ case AUD_FMT_U8:
+ return "U8";
+
+ case AUD_FMT_U16:
+ return "U16";
+
+ case AUD_FMT_S8:
+ return "S8";
+
+ case AUD_FMT_S16:
+ return "S16";
+
+ case AUD_FMT_U32:
+ return "U32";
+
+ case AUD_FMT_S32:
+ return "S32";
+ }
+
+ dolog ("Bogus audfmt %d returning S16\n", fmt);
+ return "S16";
+}
+
+static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+ int *defaultp)
+{
+ if (!strcasecmp (s, "u8")) {
+ *defaultp = 0;
+ return AUD_FMT_U8;
+ }
+ else if (!strcasecmp (s, "u16")) {
+ *defaultp = 0;
+ return AUD_FMT_U16;
+ }
+ else if (!strcasecmp (s, "u32")) {
+ *defaultp = 0;
+ return AUD_FMT_U32;
+ }
+ else if (!strcasecmp (s, "s8")) {
+ *defaultp = 0;
+ return AUD_FMT_S8;
+ }
+ else if (!strcasecmp (s, "s16")) {
+ *defaultp = 0;
+ return AUD_FMT_S16;
+ }
+ else if (!strcasecmp (s, "s32")) {
+ *defaultp = 0;
+ return AUD_FMT_S32;
+ }
+ else {
+ dolog ("Bogus audio format `%s' using %s\n",
+ s, audio_audfmt_to_string (defval));
+ *defaultp = 1;
+ return defval;
+ }
+}
+
+static audfmt_e audio_get_conf_fmt (const char *envname,
+ audfmt_e defval,
+ int *defaultp)
+{
+ const char *var = getenv (envname);
+ if (!var) {
+ *defaultp = 1;
+ return defval;
+ }
+ return audio_string_to_audfmt (var, defval, defaultp);
+}
+
+static int audio_get_conf_int (const char *key, int defval, int *defaultp)
+{
+ int val;
+ char *strval;
+
+ strval = getenv (key);
+ if (strval) {
+ *defaultp = 0;
+ val = atoi (strval);
+ return val;
+ }
+ else {
+ *defaultp = 1;
+ return defval;
+ }
+}
+
+static const char *audio_get_conf_str (const char *key,
+ const char *defval,
+ int *defaultp)
+{
+ const char *val = getenv (key);
+ if (!val) {
+ *defaultp = 1;
+ return defval;
+ }
+ else {
+ *defaultp = 0;
+ return val;
+ }
+}
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap)
+{
+ if (cap) {
+ fprintf(stderr, "%s: ", cap);
+ }
+
+ vfprintf(stderr, fmt, ap);
+}
+
+void AUD_log (const char *cap, const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (cap, fmt, ap);
+ va_end (ap);
+}
+
+static void audio_print_options (const char *prefix,
+ struct audio_option *opt)
+{
+ char *uprefix;
+
+ if (!prefix) {
+ dolog ("No prefix specified\n");
+ return;
+ }
+
+ if (!opt) {
+ dolog ("No options\n");
+ return;
+ }
+
+ uprefix = audio_alloc_prefix (prefix);
+
+ for (; opt->name; opt++) {
+ const char *state = "default";
+ printf (" %s_%s: ", uprefix, opt->name);
+
+ if (opt->overriddenp && *opt->overriddenp) {
+ state = "current";
+ }
+
+ switch (opt->tag) {
+ case AUD_OPT_BOOL:
+ {
+ int *intp = opt->valp;
+ printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
+ }
+ break;
+
+ case AUD_OPT_INT:
+ {
+ int *intp = opt->valp;
+ printf ("integer, %s = %d\n", state, *intp);
+ }
+ break;
+
+ case AUD_OPT_FMT:
+ {
+ audfmt_e *fmtp = opt->valp;
+ printf (
+ "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
+ state,
+ audio_audfmt_to_string (*fmtp)
+ );
+ }
+ break;
+
+ case AUD_OPT_STR:
+ {
+ const char **strp = opt->valp;
+ printf ("string, %s = %s\n",
+ state,
+ *strp ? *strp : "(not set)");
+ }
+ break;
+
+ default:
+ printf ("???\n");
+ dolog ("Bad value tag for option %s_%s %d\n",
+ uprefix, opt->name, opt->tag);
+ break;
+ }
+ printf (" %s\n", opt->descr);
+ }
+
+ g_free (uprefix);
+}
+
+static void audio_process_options (const char *prefix,
+ struct audio_option *opt)
+{
+ char *optname;
+ const char qemu_prefix[] = "QEMU_";
+ size_t preflen, optlen;
+
+ if (audio_bug (AUDIO_FUNC, !prefix)) {
+ dolog ("prefix = NULL\n");
+ return;
+ }
+
+ if (audio_bug (AUDIO_FUNC, !opt)) {
+ dolog ("opt = NULL\n");
+ return;
+ }
+
+ preflen = strlen (prefix);
+
+ for (; opt->name; opt++) {
+ size_t len, i;
+ int def;
+
+ if (!opt->valp) {
+ dolog ("Option value pointer for `%s' is not set\n",
+ opt->name);
+ continue;
+ }
+
+ len = strlen (opt->name);
+ /* len of opt->name + len of prefix + size of qemu_prefix
+ * (includes trailing zero) + zero + underscore (on behalf of
+ * sizeof) */
+ optlen = len + preflen + sizeof (qemu_prefix) + 1;
+ optname = g_malloc (optlen);
+
+ pstrcpy (optname, optlen, qemu_prefix);
+
+ /* copy while upper-casing, including trailing zero */
+ for (i = 0; i <= preflen; ++i) {
+ optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
+ }
+ pstrcat (optname, optlen, "_");
+ pstrcat (optname, optlen, opt->name);
+
+ def = 1;
+ switch (opt->tag) {
+ case AUD_OPT_BOOL:
+ case AUD_OPT_INT:
+ {
+ int *intp = opt->valp;
+ *intp = audio_get_conf_int (optname, *intp, &def);
+ }
+ break;
+
+ case AUD_OPT_FMT:
+ {
+ audfmt_e *fmtp = opt->valp;
+ *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
+ }
+ break;
+
+ case AUD_OPT_STR:
+ {
+ const char **strp = opt->valp;
+ *strp = audio_get_conf_str (optname, *strp, &def);
+ }
+ break;
+
+ default:
+ dolog ("Bad value tag for option `%s' - %d\n",
+ optname, opt->tag);
+ break;
+ }
+
+ if (!opt->overriddenp) {
+ opt->overriddenp = &opt->overridden;
+ }
+ *opt->overriddenp = !def;
+ g_free (optname);
+ }
+}
+
+static void audio_print_settings (struct audsettings *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;
+ case AUD_FMT_S32:
+ AUD_log (NULL, "S32");
+ break;
+ case AUD_FMT_U32:
+ AUD_log (NULL, "U32");
+ break;
+ default:
+ AUD_log (NULL, "invalid(%d)", as->fmt);
+ break;
+ }
+
+ AUD_log (NULL, " endianness=");
+ switch (as->endianness) {
+ case 0:
+ AUD_log (NULL, "little");
+ break;
+ case 1:
+ AUD_log (NULL, "big");
+ break;
+ default:
+ AUD_log (NULL, "invalid");
+ break;
+ }
+ AUD_log (NULL, "\n");
+}
+
+static int audio_validate_settings (struct audsettings *as)
+{
+ int invalid;
+
+ invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid |= as->endianness != 0 && as->endianness != 1;
+
+ switch (as->fmt) {
+ case AUD_FMT_S8:
+ case AUD_FMT_U8:
+ case AUD_FMT_S16:
+ case AUD_FMT_U16:
+ case AUD_FMT_S32:
+ case AUD_FMT_U32:
+ break;
+ default:
+ invalid = 1;
+ break;
+ }
+
+ invalid |= as->freq <= 0;
+ return invalid ? -1 : 0;
+}
+
+static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
+{
+ int bits = 8, sign = 0;
+
+ switch (as->fmt) {
+ case AUD_FMT_S8:
+ sign = 1;
+ /* fall through */
+ case AUD_FMT_U8:
+ break;
+
+ case AUD_FMT_S16:
+ sign = 1;
+ /* fall through */
+ case AUD_FMT_U16:
+ bits = 16;
+ break;
+
+ case AUD_FMT_S32:
+ sign = 1;
+ /* fall through */
+ case AUD_FMT_U32:
+ bits = 32;
+ break;
+ }
+ return info->freq == as->freq
+ && info->nchannels == as->nchannels
+ && info->sign == sign
+ && info->bits == bits
+ && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
+{
+ int bits = 8, sign = 0, shift = 0;
+
+ switch (as->fmt) {
+ case AUD_FMT_S8:
+ sign = 1;
+ case AUD_FMT_U8:
+ break;
+
+ case AUD_FMT_S16:
+ sign = 1;
+ case AUD_FMT_U16:
+ bits = 16;
+ shift = 1;
+ break;
+
+ case AUD_FMT_S32:
+ sign = 1;
+ case AUD_FMT_U32:
+ bits = 32;
+ shift = 2;
+ break;
+ }
+
+ info->freq = as->freq;
+ info->bits = bits;
+ info->sign = sign;
+ info->nchannels = as->nchannels;
+ info->shift = (as->nchannels == 2) + shift;
+ info->align = (1 << info->shift) - 1;
+ info->bytes_per_second = info->freq << info->shift;
+ info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
+}
+
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
+{
+ if (!len) {
+ return;
+ }
+
+ if (info->sign) {
+ memset (buf, 0x00, len << info->shift);
+ }
+ else {
+ switch (info->bits) {
+ case 8:
+ memset (buf, 0x80, len << info->shift);
+ break;
+
+ case 16:
+ {
+ int i;
+ uint16_t *p = buf;
+ int shift = info->nchannels - 1;
+ short s = INT16_MAX;
+
+ if (info->swap_endianness) {
+ s = bswap16 (s);
+ }
+
+ for (i = 0; i < len << shift; i++) {
+ p[i] = s;
+ }
+ }
+ break;
+
+ case 32:
+ {
+ int i;
+ uint32_t *p = buf;
+ int shift = info->nchannels - 1;
+ int32_t s = INT32_MAX;
+
+ if (info->swap_endianness) {
+ s = bswap32 (s);
+ }
+
+ for (i = 0; i < len << shift; i++) {
+ p[i] = s;
+ }
+ }
+ break;
+
+ default:
+ AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
+ info->bits);
+ break;
+ }
+ }
+}
+
+/*
+ * Capture
+ */
+static void noop_conv (struct st_sample *dst, const void *src, int samples)
+{
+ (void) src;
+ (void) dst;
+ (void) samples;
+}
+
+static CaptureVoiceOut *audio_pcm_capture_find_specific (
+ struct audsettings *as
+ )
+{
+ CaptureVoiceOut *cap;
+ AudioState *s = &glob_audio_state;
+
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ if (audio_pcm_info_eq (&cap->hw.info, as)) {
+ return cap;
+ }
+ }
+ return NULL;
+}
+
+static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
+{
+ struct capture_callback *cb;
+
+#ifdef DEBUG_CAPTURE
+ dolog ("notification %d sent\n", cmd);
+#endif
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ cb->ops.notify (cb->opaque, cmd);
+ }
+}
+
+static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
+{
+ if (cap->hw.enabled != enabled) {
+ audcnotification_e cmd;
+ cap->hw.enabled = enabled;
+ cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
+ audio_notify_capture (cap, cmd);
+ }
+}
+
+static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
+{
+ HWVoiceOut *hw = &cap->hw;
+ SWVoiceOut *sw;
+ int enabled = 0;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active) {
+ enabled = 1;
+ break;
+ }
+ }
+ audio_capture_maybe_changed (cap, enabled);
+}
+
+static void audio_detach_capture (HWVoiceOut *hw)
+{
+ SWVoiceCap *sc = hw->cap_head.lh_first;
+
+ while (sc) {
+ SWVoiceCap *sc1 = sc->entries.le_next;
+ SWVoiceOut *sw = &sc->sw;
+ CaptureVoiceOut *cap = sc->cap;
+ int was_active = sw->active;
+
+ if (sw->rate) {
+ st_rate_stop (sw->rate);
+ sw->rate = NULL;
+ }
+
+ QLIST_REMOVE (sw, entries);
+ QLIST_REMOVE (sc, entries);
+ g_free (sc);
+ if (was_active) {
+ /* We have removed soft voice from the capture:
+ this might have changed the overall status of the capture
+ since this might have been the only active voice */
+ audio_recalc_and_notify_capture (cap);
+ }
+ sc = sc1;
+ }
+}
+
+static int audio_attach_capture (HWVoiceOut *hw)
+{
+ AudioState *s = &glob_audio_state;
+ CaptureVoiceOut *cap;
+
+ audio_detach_capture (hw);
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ SWVoiceCap *sc;
+ SWVoiceOut *sw;
+ HWVoiceOut *hw_cap = &cap->hw;
+
+ sc = audio_calloc (AUDIO_FUNC, 1, sizeof (*sc));
+ if (!sc) {
+ dolog ("Could not allocate soft capture voice (%zu bytes)\n",
+ sizeof (*sc));
+ return -1;
+ }
+
+ sc->cap = cap;
+ sw = &sc->sw;
+ sw->hw = hw_cap;
+ sw->info = hw->info;
+ sw->empty = 1;
+ sw->active = hw->enabled;
+ sw->conv = noop_conv;
+ sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
+ sw->vol = nominal_volume;
+ sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
+ if (!sw->rate) {
+ dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
+ g_free (sw);
+ return -1;
+ }
+ QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
+ QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
+#ifdef DEBUG_CAPTURE
+ sw->name = g_strdup_printf ("for %p %d,%d,%d",
+ hw, sw->info.freq, sw->info.bits,
+ sw->info.nchannels);
+ dolog ("Added %s active = %d\n", sw->name, sw->active);
+#endif
+ if (sw->active) {
+ audio_capture_maybe_changed (cap, 1);
+ }
+ }
+ return 0;
+}
+
+/*
+ * Hard voice (capture)
+ */
+static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+{
+ SWVoiceIn *sw;
+ int m = hw->total_samples_captured;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active) {
+ m = audio_MIN (m, sw->total_hw_samples_acquired);
+ }
+ }
+ return m;
+}
+
+int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
+{
+ int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+ return live;
+}
+
+int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
+ int live, int pending)
+{
+ int left = hw->samples - pending;
+ int len = audio_MIN (left, live);
+ int clipped = 0;
+
+ while (len) {
+ struct st_sample *src = hw->mix_buf + hw->rpos;
+ uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
+ int samples_till_end_of_buf = hw->samples - hw->rpos;
+ int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
+
+ hw->clip (dst, src, samples_to_clip);
+
+ hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
+ len -= samples_to_clip;
+ clipped += samples_to_clip;
+ }
+ return clipped;
+}
+
+/*
+ * Soft voice (capture)
+ */
+static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
+{
+ HWVoiceIn *hw = sw->hw;
+ int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ int rpos;
+
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+
+ rpos = hw->wpos - live;
+ if (rpos >= 0) {
+ return rpos;
+ }
+ else {
+ return hw->samples + rpos;
+ }
+}
+
+int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
+{
+ HWVoiceIn *hw = sw->hw;
+ int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+ struct st_sample *src, *dst = sw->buf;
+
+ rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
+
+ live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+
+ samples = size >> sw->info.shift;
+ if (!live) {
+ return 0;
+ }
+
+ swlim = (live * sw->ratio) >> 32;
+ swlim = audio_MIN (swlim, samples);
+
+ while (swlim) {
+ src = hw->conv_buf + rpos;
+ isamp = hw->wpos - rpos;
+ /* XXX: <= ? */
+ if (isamp <= 0) {
+ isamp = hw->samples - rpos;
+ }
+
+ if (!isamp) {
+ break;
+ }
+ osamp = swlim;
+
+ if (audio_bug (AUDIO_FUNC, osamp < 0)) {
+ dolog ("osamp=%d\n", osamp);
+ return 0;
+ }
+
+ st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
+ swlim -= osamp;
+ rpos = (rpos + isamp) % hw->samples;
+ dst += osamp;
+ ret += osamp;
+ total += isamp;
+ }
+
+ if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ mixeng_volume (sw->buf, ret, &sw->vol);
+ }
+
+ sw->clip (buf, sw->buf, ret);
+ sw->total_hw_samples_acquired += total;
+ return ret << sw->info.shift;
+}
+
+/*
+ * Hard voice (playback)
+ */
+static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
+{
+ SWVoiceOut *sw;
+ int m = INT_MAX;
+ int nb_live = 0;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active || !sw->empty) {
+ m = audio_MIN (m, sw->total_hw_samples_mixed);
+ nb_live += 1;
+ }
+ }
+
+ *nb_livep = nb_live;
+ return m;
+}
+
+static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
+{
+ int smin;
+ int nb_live1;
+
+ smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
+ if (nb_live) {
+ *nb_live = nb_live1;
+ }
+
+ if (nb_live1) {
+ int live = smin;
+
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+ return live;
+ }
+ return 0;
+}
+
+/*
+ * Soft voice (playback)
+ */
+int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
+{
+ int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+ int ret = 0, pos = 0, total = 0;
+
+ if (!sw) {
+ return size;
+ }
+
+ hwsamples = sw->hw->samples;
+
+ live = sw->total_hw_samples_mixed;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
+ dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ return 0;
+ }
+
+ if (live == hwsamples) {
+#ifdef DEBUG_OUT
+ dolog ("%s is full %d\n", sw->name, live);
+#endif
+ return 0;
+ }
+
+ wpos = (sw->hw->rpos + live) % hwsamples;
+ samples = size >> sw->info.shift;
+
+ dead = hwsamples - live;
+ swlim = ((int64_t) dead << 32) / sw->ratio;
+ swlim = audio_MIN (swlim, samples);
+ if (swlim) {
+ sw->conv (sw->buf, buf, swlim);
+
+ if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ mixeng_volume (sw->buf, swlim, &sw->vol);
+ }
+ }
+
+ while (swlim) {
+ dead = hwsamples - live;
+ left = hwsamples - wpos;
+ blck = audio_MIN (dead, left);
+ if (!blck) {
+ break;
+ }
+ isamp = swlim;
+ osamp = blck;
+ st_rate_flow_mix (
+ sw->rate,
+ sw->buf + pos,
+ sw->hw->mix_buf + wpos,
+ &isamp,
+ &osamp
+ );
+ ret += isamp;
+ swlim -= isamp;
+ pos += isamp;
+ live += osamp;
+ wpos = (wpos + osamp) % hwsamples;
+ total += osamp;
+ }
+
+ sw->total_hw_samples_mixed += total;
+ sw->empty = sw->total_hw_samples_mixed == 0;
+
+#ifdef DEBUG_OUT
+ dolog (
+ "%s: write size %d ret %d total sw %d\n",
+ SW_NAME (sw),
+ size >> sw->info.shift,
+ ret,
+ sw->total_hw_samples_mixed
+ );
+#endif
+
+ return ret << sw->info.shift;
+}
+
+#ifdef DEBUG_AUDIO
+static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
+{
+ dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
+ cap, info->bits, info->sign, info->freq, info->nchannels);
+}
+#endif
+
+#define DAC
+#include "audio_template.h"
+#undef DAC
+#include "audio_template.h"
+
+/*
+ * Timer
+ */
+static int audio_is_timer_needed (void)
+{
+ HWVoiceIn *hwi = NULL;
+ HWVoiceOut *hwo = NULL;
+
+ while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ if (!hwo->poll_mode) return 1;
+ }
+ while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ if (!hwi->poll_mode) return 1;
+ }
+ return 0;
+}
+
+static void audio_reset_timer (AudioState *s)
+{
+ if (audio_is_timer_needed ()) {
+ timer_mod (s->ts,
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ }
+ else {
+ timer_del (s->ts);
+ }
+}
+
+static void audio_timer (void *opaque)
+{
+ audio_run ("timer");
+ audio_reset_timer (opaque);
+}
+
+/*
+ * Public API
+ */
+int AUD_write (SWVoiceOut *sw, void *buf, int size)
+{
+ int bytes;
+
+ if (!sw) {
+ /* XXX: Consider options */
+ return size;
+ }
+
+ if (!sw->hw->enabled) {
+ dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
+ return 0;
+ }
+
+ bytes = sw->hw->pcm_ops->write (sw, buf, size);
+ return bytes;
+}
+
+int AUD_read (SWVoiceIn *sw, void *buf, int size)
+{
+ int bytes;
+
+ if (!sw) {
+ /* XXX: Consider options */
+ return size;
+ }
+
+ if (!sw->hw->enabled) {
+ dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
+ return 0;
+ }
+
+ bytes = sw->hw->pcm_ops->read (sw, buf, size);
+ return bytes;
+}
+
+int AUD_get_buffer_size_out (SWVoiceOut *sw)
+{
+ return sw->hw->samples << sw->hw->info.shift;
+}
+
+void AUD_set_active_out (SWVoiceOut *sw, int on)
+{
+ HWVoiceOut *hw;
+
+ if (!sw) {
+ return;
+ }
+
+ hw = sw->hw;
+ if (sw->active != on) {
+ AudioState *s = &glob_audio_state;
+ SWVoiceOut *temp_sw;
+ SWVoiceCap *sc;
+
+ if (on) {
+ hw->pending_disable = 0;
+ if (!hw->enabled) {
+ hw->enabled = 1;
+ if (s->vm_running) {
+ hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
+ audio_reset_timer (s);
+ }
+ }
+ }
+ else {
+ if (hw->enabled) {
+ int nb_active = 0;
+
+ for (temp_sw = hw->sw_head.lh_first; temp_sw;
+ temp_sw = temp_sw->entries.le_next) {
+ nb_active += temp_sw->active != 0;
+ }
+
+ hw->pending_disable = nb_active == 1;
+ }
+ }
+
+ for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+ sc->sw.active = hw->enabled;
+ if (hw->enabled) {
+ audio_capture_maybe_changed (sc->cap, 1);
+ }
+ }
+ sw->active = on;
+ }
+}
+
+void AUD_set_active_in (SWVoiceIn *sw, int on)
+{
+ HWVoiceIn *hw;
+
+ if (!sw) {
+ return;
+ }
+
+ hw = sw->hw;
+ if (sw->active != on) {
+ AudioState *s = &glob_audio_state;
+ SWVoiceIn *temp_sw;
+
+ if (on) {
+ if (!hw->enabled) {
+ hw->enabled = 1;
+ if (s->vm_running) {
+ hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
+ audio_reset_timer (s);
+ }
+ }
+ sw->total_hw_samples_acquired = hw->total_samples_captured;
+ }
+ else {
+ if (hw->enabled) {
+ int nb_active = 0;
+
+ for (temp_sw = hw->sw_head.lh_first; temp_sw;
+ temp_sw = temp_sw->entries.le_next) {
+ nb_active += temp_sw->active != 0;
+ }
+
+ if (nb_active == 1) {
+ hw->enabled = 0;
+ hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
+ }
+ }
+ }
+ sw->active = on;
+ }
+}
+
+static int audio_get_avail (SWVoiceIn *sw)
+{
+ int live;
+
+ if (!sw) {
+ return 0;
+ }
+
+ live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+ 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;
+ }
+
+ ldebug (
+ "%s: get_avail live %d ret %" PRId64 "\n",
+ SW_NAME (sw),
+ live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+ );
+
+ return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+}
+
+static int audio_get_free (SWVoiceOut *sw)
+{
+ int live, dead;
+
+ if (!sw) {
+ return 0;
+ }
+
+ live = sw->total_hw_samples_mixed;
+
+ 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 %" PRId64 "\n",
+ 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_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
+{
+ int n;
+
+ if (hw->enabled) {
+ SWVoiceCap *sc;
+
+ for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+ SWVoiceOut *sw = &sc->sw;
+ int rpos2 = rpos;
+
+ n = samples;
+ while (n) {
+ int till_end_of_hw = hw->samples - rpos2;
+ int to_write = audio_MIN (till_end_of_hw, n);
+ int bytes = to_write << hw->info.shift;
+ int written;
+
+ sw->buf = hw->mix_buf + rpos2;
+ written = audio_pcm_sw_write (sw, NULL, bytes);
+ if (written - bytes) {
+ dolog ("Could not mix %d bytes into a capture "
+ "buffer, mixed %d\n",
+ bytes, written);
+ break;
+ }
+ n -= to_write;
+ rpos2 = (rpos2 + to_write) % hw->samples;
+ }
+ }
+ }
+
+ n = audio_MIN (samples, hw->samples - rpos);
+ mixeng_clear (hw->mix_buf + rpos, n);
+ mixeng_clear (hw->mix_buf, samples - n);
+}
+
+static void audio_run_out (AudioState *s)
+{
+ HWVoiceOut *hw = NULL;
+ SWVoiceOut *sw;
+
+ while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
+ int played;
+ int live, free, nb_live, cleanup_required, prev_rpos;
+
+ live = audio_pcm_hw_get_live_out (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) {
+ SWVoiceCap *sc;
+#ifdef DEBUG_OUT
+ dolog ("Disabling voice\n");
+#endif
+ hw->enabled = 0;
+ hw->pending_disable = 0;
+ hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+ for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+ sc->sw.active = 0;
+ audio_recalc_and_notify_capture (sc->cap);
+ }
+ continue;
+ }
+
+ if (!live) {
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active) {
+ free = audio_get_free (sw);
+ if (free > 0) {
+ sw->callback.fn (sw->callback.opaque, free);
+ }
+ }
+ }
+ continue;
+ }
+
+ prev_rpos = hw->rpos;
+ played = hw->pcm_ops->run_out (hw, live);
+ if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
+ dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
+ hw->rpos, hw->samples, played);
+ hw->rpos = 0;
+ }
+
+#ifdef DEBUG_OUT
+ dolog ("played=%d\n", played);
+#endif
+
+ if (played) {
+ hw->ts_helper += played;
+ audio_capture_mix_and_clear (hw, prev_rpos, played);
+ }
+
+ cleanup_required = 0;
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (!sw->active && sw->empty) {
+ continue;
+ }
+
+ if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
+ dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
+ played, sw->total_hw_samples_mixed);
+ played = sw->total_hw_samples_mixed;
+ }
+
+ sw->total_hw_samples_mixed -= played;
+
+ if (!sw->total_hw_samples_mixed) {
+ sw->empty = 1;
+ cleanup_required |= !sw->active && !sw->callback.fn;
+ }
+
+ if (sw->active) {
+ free = audio_get_free (sw);
+ if (free > 0) {
+ sw->callback.fn (sw->callback.opaque, free);
+ }
+ }
+ }
+
+ if (cleanup_required) {
+ SWVoiceOut *sw1;
+
+ sw = hw->sw_head.lh_first;
+ while (sw) {
+ sw1 = sw->entries.le_next;
+ if (!sw->active && !sw->callback.fn) {
+ audio_close_out (sw);
+ }
+ sw = sw1;
+ }
+ }
+ }
+}
+
+static void audio_run_in (AudioState *s)
+{
+ HWVoiceIn *hw = NULL;
+
+ while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
+ SWVoiceIn *sw;
+ int captured, min;
+
+ captured = hw->pcm_ops->run_in (hw);
+
+ min = audio_pcm_hw_find_min_in (hw);
+ hw->total_samples_captured += captured - min;
+ hw->ts_helper += captured;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ sw->total_hw_samples_acquired -= min;
+
+ if (sw->active) {
+ int avail;
+
+ avail = audio_get_avail (sw);
+ if (avail > 0) {
+ sw->callback.fn (sw->callback.opaque, avail);
+ }
+ }
+ }
+ }
+}
+
+static void audio_run_capture (AudioState *s)
+{
+ CaptureVoiceOut *cap;
+
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ int live, rpos, captured;
+ HWVoiceOut *hw = &cap->hw;
+ SWVoiceOut *sw;
+
+ captured = live = audio_pcm_hw_get_live_out (hw, NULL);
+ rpos = hw->rpos;
+ while (live) {
+ int left = hw->samples - rpos;
+ int to_capture = audio_MIN (live, left);
+ struct st_sample *src;
+ struct capture_callback *cb;
+
+ src = hw->mix_buf + rpos;
+ hw->clip (cap->buf, src, to_capture);
+ mixeng_clear (src, to_capture);
+
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ cb->ops.capture (cb->opaque, cap->buf,
+ to_capture << hw->info.shift);
+ }
+ rpos = (rpos + to_capture) % hw->samples;
+ live -= to_capture;
+ }
+ hw->rpos = rpos;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (!sw->active && sw->empty) {
+ continue;
+ }
+
+ if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
+ dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
+ captured, sw->total_hw_samples_mixed);
+ captured = sw->total_hw_samples_mixed;
+ }
+
+ sw->total_hw_samples_mixed -= captured;
+ sw->empty = sw->total_hw_samples_mixed == 0;
+ }
+ }
+}
+
+void audio_run (const char *msg)
+{
+ AudioState *s = &glob_audio_state;
+
+ audio_run_out (s);
+ audio_run_in (s);
+ audio_run_capture (s);
+#ifdef DEBUG_POLL
+ {
+ static double prevtime;
+ double currtime;
+ struct timeval tv;
+
+ if (gettimeofday (&tv, NULL)) {
+ perror ("audio_run: gettimeofday");
+ return;
+ }
+
+ currtime = tv.tv_sec + tv.tv_usec * 1e-6;
+ dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
+ prevtime = currtime;
+ }
+#endif
+}
+
+static struct audio_option audio_options[] = {
+ /* DAC */
+ {
+ .name = "DAC_FIXED_SETTINGS",
+ .tag = AUD_OPT_BOOL,
+ .valp = &conf.fixed_out.enabled,
+ .descr = "Use fixed settings for host DAC"
+ },
+ {
+ .name = "DAC_FIXED_FREQ",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_out.settings.freq,
+ .descr = "Frequency for fixed host DAC"
+ },
+ {
+ .name = "DAC_FIXED_FMT",
+ .tag = AUD_OPT_FMT,
+ .valp = &conf.fixed_out.settings.fmt,
+ .descr = "Format for fixed host DAC"
+ },
+ {
+ .name = "DAC_FIXED_CHANNELS",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_out.settings.nchannels,
+ .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
+ },
+ {
+ .name = "DAC_VOICES",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_out.nb_voices,
+ .descr = "Number of voices for DAC"
+ },
+ {
+ .name = "DAC_TRY_POLL",
+ .tag = AUD_OPT_BOOL,
+ .valp = &conf.try_poll_out,
+ .descr = "Attempt using poll mode for DAC"
+ },
+ /* ADC */
+ {
+ .name = "ADC_FIXED_SETTINGS",
+ .tag = AUD_OPT_BOOL,
+ .valp = &conf.fixed_in.enabled,
+ .descr = "Use fixed settings for host ADC"
+ },
+ {
+ .name = "ADC_FIXED_FREQ",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_in.settings.freq,
+ .descr = "Frequency for fixed host ADC"
+ },
+ {
+ .name = "ADC_FIXED_FMT",
+ .tag = AUD_OPT_FMT,
+ .valp = &conf.fixed_in.settings.fmt,
+ .descr = "Format for fixed host ADC"
+ },
+ {
+ .name = "ADC_FIXED_CHANNELS",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_in.settings.nchannels,
+ .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
+ },
+ {
+ .name = "ADC_VOICES",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.fixed_in.nb_voices,
+ .descr = "Number of voices for ADC"
+ },
+ {
+ .name = "ADC_TRY_POLL",
+ .tag = AUD_OPT_BOOL,
+ .valp = &conf.try_poll_in,
+ .descr = "Attempt using poll mode for ADC"
+ },
+ /* Misc */
+ {
+ .name = "TIMER_PERIOD",
+ .tag = AUD_OPT_INT,
+ .valp = &conf.period.hertz,
+ .descr = "Timer period in HZ (0 - use lowest possible)"
+ },
+ { /* End of list */ }
+};
+
+static void audio_pp_nb_voices (const char *typ, int nb)
+{
+ switch (nb) {
+ case 0:
+ printf ("Does not support %s\n", typ);
+ break;
+ case 1:
+ printf ("One %s voice\n", typ);
+ break;
+ case INT_MAX:
+ printf ("Theoretically supports many %s voices\n", typ);
+ break;
+ default:
+ printf ("Theoretically supports up to %d %s voices\n", nb, typ);
+ break;
+ }
+
+}
+
+void AUD_help (void)
+{
+ size_t i;
+
+ audio_process_options ("AUDIO", audio_options);
+ for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+ struct audio_driver *d = drvtab[i];
+ if (d->options) {
+ audio_process_options (d->name, d->options);
+ }
+ }
+
+ printf ("Audio options:\n");
+ audio_print_options ("AUDIO", audio_options);
+ printf ("\n");
+
+ printf ("Available drivers:\n");
+
+ for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+ struct audio_driver *d = drvtab[i];
+
+ printf ("Name: %s\n", d->name);
+ printf ("Description: %s\n", d->descr);
+
+ audio_pp_nb_voices ("playback", d->max_voices_out);
+ audio_pp_nb_voices ("capture", d->max_voices_in);
+
+ if (d->options) {
+ printf ("Options:\n");
+ audio_print_options (d->name, d->options);
+ }
+ else {
+ printf ("No options\n");
+ }
+ printf ("\n");
+ }
+
+ printf (
+ "Options are settable through environment variables.\n"
+ "Example:\n"
+#ifdef _WIN32
+ " set QEMU_AUDIO_DRV=wav\n"
+ " set QEMU_WAV_PATH=c:\\tune.wav\n"
+#else
+ " export QEMU_AUDIO_DRV=wav\n"
+ " export QEMU_WAV_PATH=$HOME/tune.wav\n"
+ "(for csh replace export with setenv in the above)\n"
+#endif
+ " qemu ...\n\n"
+ );
+}
+
+static int audio_driver_init (AudioState *s, struct audio_driver *drv)
+{
+ if (drv->options) {
+ audio_process_options (drv->name, drv->options);
+ }
+ s->drv_opaque = drv->init ();
+
+ if (s->drv_opaque) {
+ audio_init_nb_voices_out (drv);
+ audio_init_nb_voices_in (drv);
+ s->drv = drv;
+ return 0;
+ }
+ else {
+ dolog ("Could not init `%s' audio driver\n", drv->name);
+ return -1;
+ }
+}
+
+static void audio_vm_change_state_handler (void *opaque, int running,
+ RunState state)
+{
+ AudioState *s = opaque;
+ HWVoiceOut *hwo = NULL;
+ HWVoiceIn *hwi = NULL;
+ int op = running ? VOICE_ENABLE : VOICE_DISABLE;
+
+ s->vm_running = running;
+ while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
+ }
+
+ while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
+ }
+ audio_reset_timer (s);
+}
+
+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))) {
+ SWVoiceCap *sc;
+
+ if (hwo->enabled) {
+ hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+ }
+ hwo->pcm_ops->fini_out (hwo);
+
+ for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
+ CaptureVoiceOut *cap = sc->cap;
+ struct capture_callback *cb;
+
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ cb->ops.destroy (cb->opaque);
+ }
+ }
+ }
+
+ while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ if (hwi->enabled) {
+ hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+ }
+ hwi->pcm_ops->fini_in (hwi);
+ }
+
+ if (s->drv) {
+ s->drv->fini (s->drv_opaque);
+ }
+}
+
+static const VMStateDescription vmstate_audio = {
+ .name = "audio",
+ .version_id = 1,
+ .minimum_version_id = 1,
+ .fields = (VMStateField[]) {
+ VMSTATE_END_OF_LIST()
+ }
+};
+
+static void audio_init (void)
+{
+ size_t i;
+ int done = 0;
+ const char *drvname;
+ VMChangeStateEntry *e;
+ AudioState *s = &glob_audio_state;
+
+ if (s->drv) {
+ return;
+ }
+
+ QLIST_INIT (&s->hw_head_out);
+ QLIST_INIT (&s->hw_head_in);
+ QLIST_INIT (&s->cap_head);
+ atexit (audio_atexit);
+
+ s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
+ if (!s->ts) {
+ hw_error("Could not create audio timer\n");
+ }
+
+ 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 playback voices %d, setting to 1\n",
+ s->nb_hw_voices_out);
+ s->nb_hw_voices_out = 1;
+ }
+
+ if (s->nb_hw_voices_in <= 0) {
+ dolog ("Bogus number of capture voices %d, setting to 0\n",
+ s->nb_hw_voices_in);
+ s->nb_hw_voices_in = 0;
+ }
+
+ {
+ int def;
+ drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
+ }
+
+ if (drvname) {
+ int found = 0;
+
+ for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+ if (!strcmp (drvname, drvtab[i]->name)) {
+ done = !audio_driver_init (s, drvtab[i]);
+ found = 1;
+ break;
+ }
+ }
+
+ if (!found) {
+ dolog ("Unknown audio driver `%s'\n", drvname);
+ dolog ("Run with -audio-help to list available drivers\n");
+ }
+ }
+
+ if (!done) {
+ for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
+ if (drvtab[i]->can_be_default) {
+ done = !audio_driver_init (s, drvtab[i]);
+ }
+ }
+ }
+
+ if (!done) {
+ done = !audio_driver_init (s, &no_audio_driver);
+ if (!done) {
+ hw_error("Could not initialize audio subsystem\n");
+ }
+ else {
+ dolog ("warning: Using timer based audio emulation\n");
+ }
+ }
+
+ if (conf.period.hertz <= 0) {
+ if (conf.period.hertz < 0) {
+ dolog ("warning: Timer period is negative - %d "
+ "treating as zero\n",
+ conf.period.hertz);
+ }
+ conf.period.ticks = 1;
+ } else {
+ conf.period.ticks =
+ muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
+ }
+
+ e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
+ if (!e) {
+ dolog ("warning: Could not register change state handler\n"
+ "(Audio can continue looping even after stopping the VM)\n");
+ }
+
+ QLIST_INIT (&s->card_head);
+ vmstate_register (NULL, 0, &vmstate_audio, s);
+}
+
+void AUD_register_card (const char *name, QEMUSoundCard *card)
+{
+ audio_init ();
+ card->name = g_strdup (name);
+ memset (&card->entries, 0, sizeof (card->entries));
+ QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
+}
+
+void AUD_remove_card (QEMUSoundCard *card)
+{
+ QLIST_REMOVE (card, entries);
+ g_free (card->name);
+}
+
+
+CaptureVoiceOut *AUD_add_capture (
+ struct audsettings *as,
+ struct audio_capture_ops *ops,
+ void *cb_opaque
+ )
+{
+ AudioState *s = &glob_audio_state;
+ CaptureVoiceOut *cap;
+ struct capture_callback *cb;
+
+ if (audio_validate_settings (as)) {
+ dolog ("Invalid settings were passed when trying to add capture\n");
+ audio_print_settings (as);
+ goto err0;
+ }
+
+ cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
+ if (!cb) {
+ dolog ("Could not allocate capture callback information, size %zu\n",
+ sizeof (*cb));
+ goto err0;
+ }
+ cb->ops = *ops;
+ cb->opaque = cb_opaque;
+
+ cap = audio_pcm_capture_find_specific (as);
+ if (cap) {
+ QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+ return cap;
+ }
+ else {
+ HWVoiceOut *hw;
+ CaptureVoiceOut *cap;
+
+ cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
+ if (!cap) {
+ dolog ("Could not allocate capture voice, size %zu\n",
+ sizeof (*cap));
+ goto err1;
+ }
+
+ hw = &cap->hw;
+ QLIST_INIT (&hw->sw_head);
+ QLIST_INIT (&cap->cb_head);
+
+ /* XXX find a more elegant way */
+ hw->samples = 4096 * 4;
+ hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
+ sizeof (struct st_sample));
+ if (!hw->mix_buf) {
+ dolog ("Could not allocate capture mix buffer (%d samples)\n",
+ hw->samples);
+ goto err2;
+ }
+
+ audio_pcm_init_info (&hw->info, as);
+
+ cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+ if (!cap->buf) {
+ dolog ("Could not allocate capture buffer "
+ "(%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ goto err3;
+ }
+
+ hw->clip = mixeng_clip
+ [hw->info.nchannels == 2]
+ [hw->info.sign]
+ [hw->info.swap_endianness]
+ [audio_bits_to_index (hw->info.bits)];
+
+ QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
+ QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+
+ hw = NULL;
+ while ((hw = audio_pcm_hw_find_any_out (hw))) {
+ audio_attach_capture (hw);
+ }
+ return cap;
+
+ err3:
+ g_free (cap->hw.mix_buf);
+ err2:
+ g_free (cap);
+ err1:
+ g_free (cb);
+ err0:
+ return NULL;
+ }
+}
+
+void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
+{
+ struct capture_callback *cb;
+
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ if (cb->opaque == cb_opaque) {
+ cb->ops.destroy (cb_opaque);
+ QLIST_REMOVE (cb, entries);
+ g_free (cb);
+
+ if (!cap->cb_head.lh_first) {
+ SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
+
+ while (sw) {
+ SWVoiceCap *sc = (SWVoiceCap *) sw;
+#ifdef DEBUG_CAPTURE
+ dolog ("freeing %s\n", sw->name);
+#endif
+
+ sw1 = sw->entries.le_next;
+ if (sw->rate) {
+ st_rate_stop (sw->rate);
+ sw->rate = NULL;
+ }
+ QLIST_REMOVE (sw, entries);
+ QLIST_REMOVE (sc, entries);
+ g_free (sc);
+ sw = sw1;
+ }
+ QLIST_REMOVE (cap, entries);
+ g_free (cap);
+ }
+ return;
+ }
+ }
+}
+
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ if (sw) {
+ HWVoiceOut *hw = sw->hw;
+
+ sw->vol.mute = mute;
+ sw->vol.l = nominal_volume.l * lvol / 255;
+ sw->vol.r = nominal_volume.r * rvol / 255;
+
+ if (hw->pcm_ops->ctl_out) {
+ hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
+ }
+ }
+}
+
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ if (sw) {
+ HWVoiceIn *hw = sw->hw;
+
+ sw->vol.mute = mute;
+ sw->vol.l = nominal_volume.l * lvol / 255;
+ sw->vol.r = nominal_volume.r * rvol / 255;
+
+ if (hw->pcm_ops->ctl_in) {
+ hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
+ }
+ }
+}
Code Review / kvmfornfv.git / blobdiff