/* * 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. */ #define LOG_GROUP LOG_GROUP_DEV_AUDIO #include #include #include #include #include #include #include #include #include #include "VBoxDD.h" #include "vl_vbox.h" #include #include #define AUDIO_CAP "audio" #include "audio.h" #include "audio_int.h" #ifdef RT_OS_WINDOWS #define strcasecmp stricmp #endif /* #define DEBUG_PLIVE */ /* #define DEBUG_LIVE */ /* #define DEBUG_OUT */ /* #define DEBUG_CAPTURE */ #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown" /** * @implements PDMIAUDIOCONNECTOR */ typedef struct DRVAUDIO { /** The audio interface. */ PDMIAUDIOCONNECTOR IAudioConnector; /** Pointer to the driver instance. */ PPDMDRVINS pDrvIns; } DRVAUDIO, *PDRVAUDIO; static struct audio_driver *drvtab[] = { #if defined (RT_OS_LINUX) || defined (RT_OS_FREEBSD) || defined(VBOX_WITH_SOLARIS_OSS) &oss_audio_driver, #endif #ifdef RT_OS_LINUX # ifdef VBOX_WITH_PULSE &pulse_audio_driver, # endif # ifdef VBOX_WITH_ALSA &alsa_audio_driver, # endif #endif /* RT_OS_LINUX */ #ifdef RT_OS_FREEBSD # ifdef VBOX_WITH_PULSE &pulse_audio_driver, # endif #endif #ifdef RT_OS_DARWIN &coreaudio_audio_driver, #endif #ifdef RT_OS_WINDOWS &dsound_audio_driver, #endif #ifdef RT_OS_L4 &oss_audio_driver, #endif #ifdef RT_OS_SOLARIS &solaudio_audio_driver, #endif &no_audio_driver }; static char *audio_streamname; const char *audio_get_stream_name(void) { return audio_streamname; } struct fixed_settings { int enabled; int nb_voices; int greedy; audsettings_t settings; }; 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 */ } }, { /* ADC fixed settings */ 1, /* enabled */ 1, /* nb_voices */ 1, /* greedy */ { 44100, /* freq */ 2, /* nchannels */ AUD_FMT_S16 /* fmt */ } }, { 200 }, /* frequency (in Hz) */ 0, /* plive */ }; static AudioState glob_audio_state; volume_t nominal_volume = { 0, #ifdef FLOAT_MIXENG 1.0, 1.0 #else #ifndef VBOX UINT_MAX, UINT_MAX #else INT_MAX, INT_MAX #endif #endif }; #ifdef VBOX volume_t sum_out_volume = { 0, INT_MAX, INT_MAX }; volume_t master_out_volume = { 0, INT_MAX, INT_MAX }; volume_t pcm_out_volume = { 0, INT_MAX, INT_MAX }; volume_t pcm_in_volume = { 0, INT_MAX, INT_MAX }; #endif /* http://www.df.lth.se/~john_e/gems/gem002d.html */ /* http://www.multi-platforms.com/Tips/PopCount.htm */ uint32_t popcount (uint32_t u) { u = ((u&0x55555555) + ((u>>1)&0x55555555)); u = ((u&0x33333333) + ((u>>2)&0x33333333)); u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f)); u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff)); u = ( u&0x0000ffff) + (u>>16); return u; } uint32_t lsbindex (uint32_t u) { return popcount ((u&-u)-1); } uint64_t audio_get_clock (void) { AudioState *s; s = &glob_audio_state; return PDMDrvHlpTMGetVirtualTime (s->pDrvIns); } uint64_t audio_get_ticks_per_sec (void) { AudioState *s; s = &glob_audio_state; return PDMDrvHlpTMGetVirtualFreq (s->pDrvIns); } #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED #error No its not #else 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) { shown = 1; AUD_log (NULL, "Save all your work and restart without audio\n"); AUD_log (NULL, "Please send a bug, see www.virtualbox.org\n"); AUD_log (NULL, "I am sorry\n"); } 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=%" FMTZ "u (len=%" FMTZ "u)\n", nmemb, size, len); return NULL; } return qemu_mallocz (len); } 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_U32: return "U32"; case AUD_FMT_S8: return "S8"; case AUD_FMT_S16: return "S16"; 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 (PCFGMNODE pCfgHandle, const char *envname, audfmt_e defval, int *defaultp) { char *var = NULL; int rc; if(pCfgHandle == NULL || envname == NULL) { *defaultp = 1; return defval; } rc = CFGMR3QueryStringAlloc(pCfgHandle, envname, &var); if (RT_FAILURE (rc)) { *defaultp = 1; return defval; } return audio_string_to_audfmt (var, defval, defaultp); } static int audio_get_conf_int (PCFGMNODE pCfgHandle, const char *key, int defval, int *defaultp) { int rc; uint64_t u64Data = 0; if(pCfgHandle == NULL || key == NULL) { *defaultp = 1; return defval; } *defaultp = 0; rc = CFGMR3QueryInteger(pCfgHandle, key, &u64Data); if (RT_FAILURE (rc)) { *defaultp = 1; return defval; } else { LogFlow(("%s, Value = %d\n", key, u64Data)); *defaultp = 0; return u64Data; } } static const char *audio_get_conf_str (PCFGMNODE pCfgHandle, const char *key, const char *defval, int *defaultp) { char *val = NULL; int rc; if(pCfgHandle == NULL || key == NULL) { *defaultp = 1; return defval; } rc = CFGMR3QueryStringAlloc(pCfgHandle, key, &val); if (RT_FAILURE (rc)) { *defaultp = 1; return defval; } else { *defaultp = 0; return val; } } void AUD_vlog (const char *cap, const char *fmt, va_list va) { va_list va2; va_copy (va2, va); /* Have to make a copy here or GCC will break. */ if (cap) { Log (("%s: %N", cap, fmt, &va2)); } else { Log (("%N", fmt, &va2)); } va_end (va2); } void AUD_log (const char *cap, const char *fmt, ...) { va_list va; va_start (va, fmt); AUD_vlog (cap, fmt, va); va_end (va); } static void audio_process_options (PCFGMNODE pCfgHandle, const char *prefix, struct audio_option *opt) { int def; PCFGMNODE pCfgChildHandle = NULL; PCFGMNODE pCfgChildChildHandle = NULL; if (audio_bug (AUDIO_FUNC, !prefix)) { dolog ("prefix = NULL\n"); return; } if (audio_bug (AUDIO_FUNC, !opt)) { dolog ("opt = NULL\n"); return; } /* if pCfgHandle is NULL, let NULL be passed to get int and get string functions.. * The getter function will return default values. */ if(pCfgHandle != NULL) { /* If its audio general setting, need to traverse to one child node. * /Devices/ihac97/0/LUN#0/Config/Audio */ if(!strncmp(prefix, "AUDIO", 5)) { pCfgChildHandle = CFGMR3GetFirstChild(pCfgHandle); if(pCfgChildHandle) { pCfgHandle = pCfgChildHandle; } } else { /* If its driver specific configuration , then need to traverse two level deep child * child nodes. for eg. in case of DirectSoundConfiguration item * /Devices/ihac97/0/LUN#0/Config/Audio/DirectSoundConfig */ pCfgChildHandle = CFGMR3GetFirstChild(pCfgHandle); if (pCfgChildHandle) { pCfgChildChildHandle = CFGMR3GetFirstChild(pCfgChildHandle); if(pCfgChildChildHandle) { pCfgHandle = pCfgChildChildHandle; } } } } for (; opt->name; opt++) { if (!opt->valp) { dolog ("Option value pointer for `%s' is not set\n", opt->name); continue; } def = 1; switch (opt->tag) { case AUD_OPT_BOOL: case AUD_OPT_INT: { int *intp = opt->valp; *intp = audio_get_conf_int(pCfgHandle, opt->name, *intp, &def); } break; case AUD_OPT_FMT: { audfmt_e *fmtp = opt->valp; *fmtp = audio_get_conf_fmt (pCfgHandle, opt->name, *fmtp, &def); } break; case AUD_OPT_STR: { const char **strp = opt->valp; *strp = audio_get_conf_str (pCfgHandle, opt->name, *strp, &def); } break; default: dolog ("Bad value tag for option `%s' - %d\n", opt->name, opt->tag); break; } if (!opt->overridenp) { opt->overridenp = &opt->overriden; } *opt->overridenp = !def; } } 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; 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 (audsettings_t *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, audsettings_t *as) { int bits = 8, sign = 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; break; case AUD_FMT_S32: sign = 1; 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, audsettings_t *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 (st_sample_t *dst, const void *src, int samples, volume_t *vol) { (void) src; (void) dst; (void) samples; (void) vol; } static CaptureVoiceOut *audio_pcm_capture_find_specific ( AudioState *s, audsettings_t *as ) { CaptureVoiceOut *cap; 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; } LIST_REMOVE (sw, entries); LIST_REMOVE (sc, entries); qemu_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 (AudioState *s, HWVoiceOut *hw) { 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 (%u 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->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)); qemu_free (sw); return -1; } LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries); LIST_INSERT_HEAD (&hw->cap_head, sc, entries); #ifdef DEBUG_CAPTURE asprintf (&sw->name, "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; } /* * 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; st_sample_t *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; } if (ret + osamp > sw->buf_samples) Log(("audio_pcm_sw_read: buffer overflow!! ret = %d, osamp = %d, buf_samples = %d\n", ret, osamp, sw->buf_samples)); st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; rpos = (rpos + isamp) % hw->samples; dst += osamp; ret += osamp; total += isamp; } if (ret > sw->buf_samples) Log(("audio_pcm_sw_read: buffer overflow!! ret = %d, buf_samples = %d\n", ret, sw->buf_samples)); 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; } int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live) { int smin; smin = audio_pcm_hw_find_min_out (hw, nb_live); if (!*nb_live) { return 0; } else { 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; } } int audio_pcm_hw_get_live_out (HWVoiceOut *hw) { int nb_live; int live; live = audio_pcm_hw_get_live_out2 (hw, &nb_live); if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) { dolog ("live=%d hw->samples=%d\n", live, hw->samples); return 0; } return live; } /* * 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->buf_samples) Log(("audio_pcm_sw_write: buffer overflow!! swlim = %d, buf_samples = %d\n", swlim, pos, sw->buf_samples)); if (swlim) { #ifndef VBOX sw->conv (sw->buf, buf, swlim, &sw->vol); #else sw->conv (sw->buf, buf, swlim, &sum_out_volume); #endif } while (swlim) { dead = hwsamples - live; left = hwsamples - wpos; blck = audio_MIN (dead, left); if (!blck) { break; } isamp = swlim; osamp = blck; if (pos + isamp > sw->buf_samples) Log(("audio_pcm_sw_write: buffer overflow!! isamp = %d, pos = %d, buf_samples = %d\n", isamp, pos, sw->buf_samples)); 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" 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) { SWVoiceOut *temp_sw; SWVoiceCap *sc; if (on) { hw->pending_disable = 0; if (!hw->enabled) { hw->enabled = 1; hw->pcm_ops->ctl_out (hw, VOICE_ENABLE); } } 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) { SWVoiceIn *temp_sw; if (on) { if (!hw->enabled) { hw->enabled = 1; hw->pcm_ops->ctl_in (hw, VOICE_ENABLE); } 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 %lld\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 %lld\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_sniff_and_clear (hw, hw->mix_buf + rpos, n); mixeng_sniff_and_clear (hw, 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 (s, hw))) { int played; int live, myfree, nb_live, cleanup_required, prev_rpos; 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) { 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) { myfree = audio_get_free (sw); if (myfree > 0) { sw->callback.fn (sw->callback.opaque, myfree); } } } continue; } prev_rpos = hw->rpos; played = hw->pcm_ops->run_out (hw); 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) { myfree = audio_get_free (sw); if (myfree > 0) { sw->callback.fn (sw->callback.opaque, myfree); } } } if (cleanup_required) { SWVoiceOut *sw1; sw = hw->sw_head.lh_first; while (sw) { sw1 = 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); } sw = sw1; } } } } static void audio_run_in (AudioState *s) { HWVoiceIn *hw = NULL; while ((hw = audio_pcm_hw_find_any_enabled_in (s, 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); rpos = hw->rpos; while (live) { int left = hw->samples - rpos; int to_capture = audio_MIN (live, left); st_sample_t *src; struct capture_callback *cb; src = hw->mix_buf + rpos; hw->clip (cap->buf, src, to_capture); mixeng_sniff_and_clear (hw, 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; } } } static void audio_timer (void *opaque) { AudioState *s = opaque; audio_run_out (s); audio_run_in (s); audio_run_capture (s); TMTimerSet (s->ts, TMTimerGet (s->ts) + conf.period.ticks); } static struct audio_option audio_options[] = { /* DAC */ {"DACFixedSettings", AUD_OPT_BOOL, &conf.fixed_out.enabled, "Use fixed settings for host DAC", NULL, 0}, {"DACFixedFreq", AUD_OPT_INT, &conf.fixed_out.settings.freq, "Frequency for fixed host DAC", NULL, 0}, {"DACFixedFmt", AUD_OPT_FMT, &conf.fixed_out.settings.fmt, "Format for fixed host DAC", NULL, 0}, {"DACFixedChannels", AUD_OPT_INT, &conf.fixed_out.settings.nchannels, "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0}, {"DACVoices", AUD_OPT_INT, &conf.fixed_out.nb_voices, "Number of voices for DAC", NULL, 0}, /* ADC */ {"ADCFixedSettings", AUD_OPT_BOOL, &conf.fixed_in.enabled, "Use fixed settings for host ADC", NULL, 0}, {"ADCFixedFreq", AUD_OPT_INT, &conf.fixed_in.settings.freq, "Frequency for fixed host ADC", NULL, 0}, {"ADCFixedFmt", AUD_OPT_FMT, &conf.fixed_in.settings.fmt, "Format for fixed host ADC", NULL, 0}, {"ADCFixedChannels", AUD_OPT_INT, &conf.fixed_in.settings.nchannels, "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0}, {"ADCVoices", AUD_OPT_INT, &conf.fixed_in.nb_voices, "Number of voices for ADC", NULL, 0}, /* Misc */ {"TimreFreq", AUD_OPT_INT, &conf.period.hz, "Timer frequency in Hz (0 - use lowest possible)", NULL, 0}, {"PLIVE", AUD_OPT_BOOL, &conf.plive, "(undocumented)", NULL, 0}, {NULL, 0, NULL, NULL, NULL, 0} }; static int audio_driver_init (PCFGMNODE pCfgHandle, AudioState *s, struct audio_driver *drv) { if (drv->options) { audio_process_options (pCfgHandle, drv->name, drv->options); } s->drv_opaque = drv->init (); if (s->drv_opaque) { /* Filter must be installed before initializing voices. */ drv = filteraudio_install(drv, s->drv_opaque); audio_init_nb_voices_out (s, drv); audio_init_nb_voices_in (s, 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) { AudioState *s = opaque; HWVoiceOut *hwo = NULL; HWVoiceIn *hwi = NULL; int op = running ? VOICE_ENABLE : VOICE_DISABLE; while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) { hwo->pcm_ops->ctl_out (hwo, op); } while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) { hwi->pcm_ops->ctl_in (hwi, op); } } static void audio_atexit (void) { AudioState *s = &glob_audio_state; HWVoiceOut *hwo = NULL; HWVoiceIn *hwi = NULL; /* VBox change: audio_pcm_hw_find_any_enabled_out => audio_pcm_hw_find_any_out */ while ((hwo = audio_pcm_hw_find_any_out (s, hwo))) { SWVoiceCap *sc; 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); } } } /* VBox change: audio_pcm_hw_find_any_enabled_in => audio_pcm_hw_find_any_in */ while ((hwi = audio_pcm_hw_find_any_in (s, hwi))) { hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE); hwi->pcm_ops->fini_in (hwi); } if (s->drv) { s->drv->fini (s->drv_opaque); } } void AUD_register_card (const char *name, QEMUSoundCard *card) { AudioState *s = &glob_audio_state; 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); } static DECLCALLBACK(void) audio_timer_helper (PPDMDRVINS pDrvIns, PTMTIMER pTimer, void *pvUser) { AudioState *s = (AudioState *)pvUser; audio_timer (s); } static int AUD_init (PCFGMNODE pCfgHandle, PPDMDRVINS pDrvIns, const char *drvname) { size_t i; int done = 0; AudioState *s = &glob_audio_state; int rc; LIST_INIT (&s->hw_head_out); LIST_INIT (&s->hw_head_in); LIST_INIT (&s->cap_head); rc = PDMDrvHlpTMTimerCreate (pDrvIns, TMCLOCK_VIRTUAL, audio_timer_helper, &glob_audio_state, 0, "Audio timer", &s->ts); if (RT_FAILURE (rc)) return rc; audio_process_options (pCfgHandle, "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; } LogRel(("Audio: Trying driver '%s'.\n", drvname)); if (drvname) { int found = 0; for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) { if (!strcmp (drvname, drvtab[i]->name)) { done = !audio_driver_init (pCfgHandle, s, drvtab[i]); found = 1; break; } } if (!found) { dolog ("Unknown audio driver `%s'\n", drvname); } } if (!done) { for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) { if (drvtab[i]->can_be_default) { LogRel(("Audio: Initialization of driver '%s' failed, trying '%s'.\n", drvname, drvtab[i]->name)); drvname = drvtab[i]->name; done = !audio_driver_init (pCfgHandle, s, drvtab[i]); } } } if (!done) { done = !audio_driver_init (pCfgHandle, s, &no_audio_driver); if (!done) { dolog ("Could not initialize audio subsystem\n"); } else { LogRel(("Audio: Initialization of driver '%s' failed, using NULL driver.\n", drvname)); dolog ("warning: Using timer based audio emulation\n"); } } 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; } else { conf.period.ticks = PDMDrvHlpTMGetVirtualFreq (pDrvIns) / conf.period.hz; } } else { /* XXX */ rc = TMR3TimerDestroy (s->ts); return rc; } LIST_INIT (&s->card_head); TMTimerSet (s->ts, TMTimerGet (s->ts) + conf.period.ticks); return VINF_SUCCESS; } int AUD_init_null(void) { AudioState *s = &glob_audio_state; #ifdef VBOX if (s->drv) s->drv->fini (s->drv_opaque); #endif LogRel(("Audio: Using NULL audio driver\n")); return audio_driver_init (NULL, s, &no_audio_driver); } CaptureVoiceOut *AUD_add_capture ( AudioState *s, audsettings_t *as, struct audio_capture_ops *ops, void *cb_opaque ) { CaptureVoiceOut *cap; struct capture_callback *cb; if (!s) { /* XXX suppress */ s = &glob_audio_state; } 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 %u\n", sizeof (*cb)); goto err0; } cb->ops = *ops; cb->opaque = cb_opaque; cap = audio_pcm_capture_find_specific (s, as); if (cap) { LIST_INSERT_HEAD (&cap->cb_head, cb, entries); return cap; } else { HWVoiceOut *hw; #ifndef VBOX CaptureVoiceOut *cap; #endif cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap)); if (!cap) { dolog ("Could not allocate capture voice, size %u\n", sizeof (*cap)); goto err1; } hw = &cap->hw; LIST_INIT (&hw->sw_head); LIST_INIT (&cap->cb_head); /* XXX find a more elegant way */ hw->samples = 4096 * 4; hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (st_sample_t)); 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)]; LIST_INSERT_HEAD (&s->cap_head, cap, entries); LIST_INSERT_HEAD (&cap->cb_head, cb, entries); hw = NULL; while ((hw = audio_pcm_hw_find_any_out (s, hw))) { audio_attach_capture (s, hw); } return cap; err3: qemu_free (cap->hw.mix_buf); err2: qemu_free (cap); err1: qemu_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); LIST_REMOVE (cb, entries); qemu_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; } LIST_REMOVE (sw, entries); LIST_REMOVE (sc, entries); qemu_free (sc); sw = sw1; } LIST_REMOVE (cap, entries); qemu_free (cap); } return; } } } void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol) { if (sw) { sw->vol.mute = mute; sw->vol.l = (uint32_t)lvol * 0x808080; /* maximum is INT_MAX = 0x7fffffff */ sw->vol.r = (uint32_t)rvol * 0x808080; /* maximum is INT_MAX = 0x7fffffff */ } } void AUD_set_volume (audmixerctl_t mt, int *mute, uint8_t *lvol, uint8_t *rvol) { volume_t *vol = NULL; const char *name; switch (mt) { case AUD_MIXER_VOLUME: name = "MASTER"; vol = &master_out_volume; break; case AUD_MIXER_PCM: name = "PCM_OUT"; vol = &pcm_out_volume; break; case AUD_MIXER_LINE_IN: name = "LINE_IN"; vol = &pcm_in_volume; break; default: return; } if (vol) { uint32_t u32VolumeLeft = (uint32_t)*lvol; uint32_t u32VolumeRight = (uint32_t)*rvol; /* 0x00..0xff => 0x01..0x100 */ if (u32VolumeLeft) u32VolumeLeft++; if (u32VolumeRight) u32VolumeRight++; vol->mute = *mute; vol->l = u32VolumeLeft * 0x800000; /* maximum is 0x80000000 */ vol->r = u32VolumeRight * 0x800000; /* maximum is 0x80000000 */ } sum_out_volume.mute = master_out_volume.mute || pcm_out_volume.mute; sum_out_volume.l = ASMMultU64ByU32DivByU32(master_out_volume.l, pcm_out_volume.l, 0x80000000U); sum_out_volume.r = ASMMultU64ByU32DivByU32(master_out_volume.r, pcm_out_volume.r, 0x80000000U); } void AUD_set_record_source (audrecsource_t *ars, audrecsource_t *als) { LogRel(("Audio: set_record_source ars=%d als=%d (not implemented)\n", *ars, *als)); } int AUD_is_host_voice_in_ok(SWVoiceIn *sw) { AudioState *s = &glob_audio_state; if (sw == NULL) { return 0; } return filteraudio_is_host_voice_in_ok(s->drv, sw->hw); } int AUD_is_host_voice_out_ok(SWVoiceOut *sw) { AudioState *s = &glob_audio_state; if (sw == NULL) { return 0; } return filteraudio_is_host_voice_out_ok(s->drv, sw->hw); } /** * @interface_method_impl{PDMIBASE,pfnQueryInterface} */ static DECLCALLBACK(void *) drvAudioQueryInterface(PPDMIBASE pInterface, const char *pszIID) { PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIAUDIOCONNECTOR, &pThis->IAudioConnector); return NULL; } /** * Power Off notification. * * @param pDrvIns The driver instance data. */ static DECLCALLBACK(void) drvAudioPowerOff(PPDMDRVINS pDrvIns) { AudioState *s = &glob_audio_state; audio_vm_change_state_handler (s, 0); } /** * Destruct a driver instance. * * Most VM resources are freed by the VM. This callback is provided so that any non-VM * resources can be freed correctly. * * @param pDrvIns The driver instance data. */ static DECLCALLBACK(void) drvAudioDestruct(PPDMDRVINS pDrvIns) { LogFlow(("drvAUDIODestruct:\n")); PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns); if (audio_streamname) { MMR3HeapFree(audio_streamname); audio_streamname = NULL; } audio_atexit (); } /** * Construct an AUDIO driver instance. * * @copydoc FNPDMDRVCONSTRUCT */ static DECLCALLBACK(int) drvAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfgHandle, uint32_t fFlags) { PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); char *drvname; int rc; LogFlow(("drvAUDIOConstruct:\n")); PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns); /* * Validate the config. */ if (!CFGMR3AreValuesValid(pCfgHandle, "AudioDriver\0StreamName\0")) return VERR_PDM_DRVINS_UNKNOWN_CFG_VALUES; /* * Init the static parts. */ pThis->pDrvIns = pDrvIns; /* IBase */ pDrvIns->IBase.pfnQueryInterface = drvAudioQueryInterface; /* IAudio */ /* pThis->IAudioConnector.pfn; */ glob_audio_state.pDrvIns = pDrvIns; rc = CFGMR3QueryStringAlloc (pCfgHandle, "AudioDriver", &drvname); if (RT_FAILURE (rc)) return rc; rc = CFGMR3QueryStringAlloc (pCfgHandle, "StreamName", &audio_streamname); if (RT_FAILURE (rc)) audio_streamname = NULL; rc = AUD_init (pCfgHandle, pDrvIns, drvname); if (RT_FAILURE (rc)) return rc; MMR3HeapFree (drvname); return VINF_SUCCESS; } /** * Suspend notification. * * @returns VBox status. * @param pDrvIns The driver instance data. */ static DECLCALLBACK(void) drvAudioSuspend(PPDMDRVINS pDrvIns) { AudioState *s = &glob_audio_state; audio_vm_change_state_handler (s, 0); } /** * Resume notification. * * @returns VBox status. * @param pDrvIns The driver instance data. */ static DECLCALLBACK(void) audioResume(PPDMDRVINS pDrvIns) { AudioState *s = &glob_audio_state; audio_vm_change_state_handler (s, 1); } /** * Audio driver registration record. */ const PDMDRVREG g_DrvAUDIO = { /* u32Version */ PDM_DRVREG_VERSION, /* szName */ "AUDIO", /* szRCMod */ "", /* szR0Mod */ "", /* pszDescription */ "AUDIO Driver", /* fFlags */ PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT, /* fClass. */ PDM_DRVREG_CLASS_AUDIO, /* cMaxInstances */ 1, /* cbInstance */ sizeof(DRVAUDIO), /* pfnConstruct */ drvAudioConstruct, /* pfnDestruct */ drvAudioDestruct, /* pfnRelocate */ NULL, /* pfnIOCtl */ NULL, /* pfnPowerOn */ NULL, /* pfnReset */ NULL, /* pfnSuspend */ drvAudioSuspend, /* pfnResume */ audioResume, /* pfnAttach */ NULL, /* pfnDetach */ NULL, /* pfnPowerOff */ drvAudioPowerOff, /* pfnSoftReset */ NULL, /* u32EndVersion */ PDM_DRVREG_VERSION };