| /* |
| * 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 "qemu/osdep.h" |
| #include "hw/hw.h" |
| #include "audio.h" |
| #include "monitor/monitor.h" |
| #include "qemu/timer.h" |
| #include "sysemu/sysemu.h" |
| #include "qemu/cutils.h" |
| #include "sysemu/replay.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 const char *audio_prio_list[] = { |
| "spice", |
| CONFIG_AUDIO_DRIVERS |
| "none", |
| "wav", |
| }; |
| |
| static QLIST_HEAD(, audio_driver) audio_drivers; |
| |
| void audio_driver_register(audio_driver *drv) |
| { |
| QLIST_INSERT_HEAD(&audio_drivers, drv, next); |
| } |
| |
| audio_driver *audio_driver_lookup(const char *name) |
| { |
| struct audio_driver *d; |
| |
| QLIST_FOREACH(d, &audio_drivers, next) { |
| if (strcmp(name, d->name) == 0) { |
| return d; |
| } |
| } |
| |
| audio_module_load_one(name); |
| QLIST_FOREACH(d, &audio_drivers, next) { |
| if (strcmp(name, d->name) == 0) { |
| return d; |
| } |
| } |
| |
| return NULL; |
| } |
| |
| static void audio_module_load_all(void) |
| { |
| int i; |
| |
| for (i = 0; i < ARRAY_SIZE(audio_prio_list); i++) { |
| audio_driver_lookup(audio_prio_list[i]); |
| } |
| } |
| |
| 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 && !qemu_strtoi(strval, NULL, 10, &val)) { |
| *defaultp = 0; |
| 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(__func__, !prefix)) { |
| dolog ("prefix = NULL\n"); |
| return; |
| } |
| |
| if (audio_bug(__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(__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(__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(__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(__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(__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(__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(__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_anticipate_ns(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) |
| { |
| if (!sw) { |
| /* XXX: Consider options */ |
| return size; |
| } |
| |
| if (!sw->hw->enabled) { |
| dolog ("Writing to disabled voice %s\n", SW_NAME (sw)); |
| return 0; |
| } |
| |
| return sw->hw->pcm_ops->write(sw, buf, size); |
| } |
| |
| int AUD_read (SWVoiceIn *sw, void *buf, int size) |
| { |
| if (!sw) { |
| /* XXX: Consider options */ |
| return size; |
| } |
| |
| if (!sw->hw->enabled) { |
| dolog ("Reading from disabled voice %s\n", SW_NAME (sw)); |
| return 0; |
| } |
| |
| return sw->hw->pcm_ops->read(sw, buf, size); |
| } |
| |
| 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(__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(__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(__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); |
| replay_audio_out(&played); |
| if (audio_bug(__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(__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 = 0, min; |
| |
| if (replay_mode != REPLAY_MODE_PLAY) { |
| captured = hw->pcm_ops->run_in(hw); |
| } |
| replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples); |
| |
| 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(__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) |
| { |
| struct audio_driver *d; |
| |
| /* make sure we print the help text for modular drivers too */ |
| audio_module_load_all(); |
| |
| audio_process_options ("AUDIO", audio_options); |
| QLIST_FOREACH(d, &audio_drivers, next) { |
| 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"); |
| |
| QLIST_FOREACH(d, &audio_drivers, next) { |
| |
| 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 bool is_cleaning_up; |
| |
| bool audio_is_cleaning_up(void) |
| { |
| return is_cleaning_up; |
| } |
| |
| void audio_cleanup(void) |
| { |
| AudioState *s = &glob_audio_state; |
| HWVoiceOut *hwo, *hwon; |
| HWVoiceIn *hwi, *hwin; |
| |
| is_cleaning_up = true; |
| QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) { |
| 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); |
| } |
| } |
| QLIST_REMOVE(hwo, entries); |
| } |
| |
| QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) { |
| if (hwi->enabled) { |
| hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE); |
| } |
| hwi->pcm_ops->fini_in (hwi); |
| QLIST_REMOVE(hwi, entries); |
| } |
| |
| if (s->drv) { |
| s->drv->fini (s->drv_opaque); |
| s->drv = NULL; |
| } |
| } |
| |
| 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; |
| struct audio_driver *driver; |
| |
| if (s->drv) { |
| return; |
| } |
| |
| QLIST_INIT (&s->hw_head_out); |
| QLIST_INIT (&s->hw_head_in); |
| QLIST_INIT (&s->cap_head); |
| atexit(audio_cleanup); |
| |
| s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s); |
| |
| 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) { |
| driver = audio_driver_lookup(drvname); |
| if (driver) { |
| done = !audio_driver_init(s, driver); |
| } else { |
| 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(audio_prio_list); i++) { |
| driver = audio_driver_lookup(audio_prio_list[i]); |
| if (driver && driver->can_be_default) { |
| done = !audio_driver_init(s, driver); |
| } |
| } |
| } |
| |
| if (!done) { |
| driver = audio_driver_lookup("none"); |
| done = !audio_driver_init(s, driver); |
| assert(done); |
| 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 = NANOSECONDS_PER_SECOND / 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(__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(__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(__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(__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); |
| |
| QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) { |
| 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->hw.mix_buf); |
| g_free (cap->buf); |
| 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); |
| } |
| } |
| } |