/* * 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 "audio.h" #include "migration/vmstate.h" #include "monitor/monitor.h" #include "qemu/timer.h" #include "qapi/error.h" #include "qapi/qobject-input-visitor.h" #include "qapi/qapi-visit-audio.h" #include "qemu/cutils.h" #include "qemu/module.h" #include "sysemu/replay.h" #include "sysemu/runstate.h" #include "ui/qemu-spice.h" #include "trace.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. */ const char *audio_prio_list[] = { "spice", CONFIG_AUDIO_DRIVERS "none", "wav", NULL }; static QLIST_HEAD(, audio_driver) audio_drivers; static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs); 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 QTAILQ_HEAD(AudioStateHead, AudioState) audio_states = QTAILQ_HEAD_INITIALIZER(audio_states); 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 }; static bool legacy_config = true; 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, "I am sorry\n"); } AUD_log (NULL, "Context:\n"); abort(); } return cond; } 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); } 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_settings (struct audsettings *as) { dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels); switch (as->fmt) { case AUDIO_FORMAT_S8: AUD_log (NULL, "S8"); break; case AUDIO_FORMAT_U8: AUD_log (NULL, "U8"); break; case AUDIO_FORMAT_S16: AUD_log (NULL, "S16"); break; case AUDIO_FORMAT_U16: AUD_log (NULL, "U16"); break; case AUDIO_FORMAT_S32: AUD_log (NULL, "S32"); break; case AUDIO_FORMAT_U32: AUD_log (NULL, "U32"); break; case AUDIO_FORMAT_F32: AUD_log (NULL, "F32"); 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; invalid |= as->endianness != 0 && as->endianness != 1; switch (as->fmt) { case AUDIO_FORMAT_S8: case AUDIO_FORMAT_U8: case AUDIO_FORMAT_S16: case AUDIO_FORMAT_U16: case AUDIO_FORMAT_S32: case AUDIO_FORMAT_U32: case AUDIO_FORMAT_F32: 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; bool is_signed = false, is_float = false; switch (as->fmt) { case AUDIO_FORMAT_S8: is_signed = true; /* fall through */ case AUDIO_FORMAT_U8: break; case AUDIO_FORMAT_S16: is_signed = true; /* fall through */ case AUDIO_FORMAT_U16: bits = 16; break; case AUDIO_FORMAT_F32: is_float = true; /* fall through */ case AUDIO_FORMAT_S32: is_signed = true; /* fall through */ case AUDIO_FORMAT_U32: bits = 32; break; default: abort(); } return info->freq == as->freq && info->nchannels == as->nchannels && info->is_signed == is_signed && info->is_float == is_float && 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, mul; bool is_signed = false, is_float = false; switch (as->fmt) { case AUDIO_FORMAT_S8: is_signed = true; /* fall through */ case AUDIO_FORMAT_U8: mul = 1; break; case AUDIO_FORMAT_S16: is_signed = true; /* fall through */ case AUDIO_FORMAT_U16: bits = 16; mul = 2; break; case AUDIO_FORMAT_F32: is_float = true; /* fall through */ case AUDIO_FORMAT_S32: is_signed = true; /* fall through */ case AUDIO_FORMAT_U32: bits = 32; mul = 4; break; default: abort(); } info->freq = as->freq; info->bits = bits; info->is_signed = is_signed; info->is_float = is_float; info->nchannels = as->nchannels; info->bytes_per_frame = as->nchannels * mul; info->bytes_per_second = info->freq * info->bytes_per_frame; 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->is_signed || info->is_float) { memset(buf, 0x00, len * info->bytes_per_frame); } else { switch (info->bits) { case 8: memset(buf, 0x80, len * info->bytes_per_frame); break; case 16: { int i; uint16_t *p = buf; short s = INT16_MAX; if (info->swap_endianness) { s = bswap16 (s); } for (i = 0; i < len * info->nchannels; i++) { p[i] = s; } } break; case 32: { int i; uint32_t *p = buf; int32_t s = INT32_MAX; if (info->swap_endianness) { s = bswap32 (s); } for (i = 0; i < len * info->nchannels; 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(AudioState *s, struct audsettings *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; } 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 = hw->s; 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 = g_malloc0(sizeof(*sc)); 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 size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) { SWVoiceIn *sw; size_t m = hw->total_samples_captured; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (sw->active) { m = MIN (m, sw->total_hw_samples_acquired); } } return m; } static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) { size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); if (audio_bug(__func__, live > hw->conv_buf->size)) { dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); return 0; } return live; } static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) { size_t clipped = 0; size_t pos = hw->mix_buf->pos; while (len) { st_sample *src = hw->mix_buf->samples + pos; uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame); size_t samples_till_end_of_buf = hw->mix_buf->size - pos; size_t samples_to_clip = MIN(len, samples_till_end_of_buf); hw->clip(dst, src, samples_to_clip); pos = (pos + samples_to_clip) % hw->mix_buf->size; len -= samples_to_clip; clipped += samples_to_clip; } } /* * Soft voice (capture) */ static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw) { HWVoiceIn *hw = sw->hw; ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired; ssize_t rpos; if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) { dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); return 0; } rpos = hw->conv_buf->pos - live; if (rpos >= 0) { return rpos; } else { return hw->conv_buf->size + rpos; } } static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) { HWVoiceIn *hw = sw->hw; size_t 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->conv_buf->size; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (audio_bug(__func__, live > hw->conv_buf->size)) { dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); return 0; } samples = size / sw->info.bytes_per_frame; if (!live) { return 0; } swlim = (live * sw->ratio) >> 32; swlim = MIN (swlim, samples); while (swlim) { src = hw->conv_buf->samples + rpos; if (hw->conv_buf->pos > rpos) { isamp = hw->conv_buf->pos - rpos; } else { isamp = hw->conv_buf->size - rpos; } if (!isamp) { break; } osamp = swlim; st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; rpos = (rpos + isamp) % hw->conv_buf->size; dst += osamp; ret += osamp; total += isamp; } if (hw->pcm_ops && !hw->pcm_ops->volume_in) { mixeng_volume (sw->buf, ret, &sw->vol); } sw->clip (buf, sw->buf, ret); sw->total_hw_samples_acquired += total; return ret * sw->info.bytes_per_frame; } /* * Hard voice (playback) */ static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep) { SWVoiceOut *sw; size_t m = SIZE_MAX; int nb_live = 0; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (sw->active || !sw->empty) { m = MIN (m, sw->total_hw_samples_mixed); nb_live += 1; } } *nb_livep = nb_live; return m; } static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) { size_t smin; int nb_live1; smin = audio_pcm_hw_find_min_out (hw, &nb_live1); if (nb_live) { *nb_live = nb_live1; } if (nb_live1) { size_t live = smin; if (audio_bug(__func__, live > hw->mix_buf->size)) { dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); return 0; } return live; } return 0; } /* * Soft voice (playback) */ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) { size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; size_t ret = 0, pos = 0, total = 0; if (!sw) { return size; } hwsamples = sw->hw->mix_buf->size; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hwsamples)) { dolog("live=%zu hw->mix_buf->size=%zu\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->mix_buf->pos + live) % hwsamples; samples = size / sw->info.bytes_per_frame; dead = hwsamples - live; swlim = ((int64_t) dead << 32) / sw->ratio; swlim = MIN (swlim, samples); if (swlim) { sw->conv (sw->buf, buf, swlim); if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) { mixeng_volume (sw->buf, swlim, &sw->vol); } } while (swlim) { dead = hwsamples - live; left = hwsamples - wpos; blck = MIN (dead, left); if (!blck) { break; } isamp = swlim; osamp = blck; st_rate_flow_mix ( sw->rate, sw->buf + pos, sw->hw->mix_buf->samples + 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 %zu ret %zu total sw %zu\n", SW_NAME (sw), size / sw->info.bytes_per_frame, ret, sw->total_hw_samples_mixed ); #endif return ret * sw->info.bytes_per_frame; } #ifdef DEBUG_AUDIO static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info) { dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n", cap, info->bits, info->is_signed, info->is_float, info->freq, info->nchannels); } #endif #define DAC #include "audio_template.h" #undef DAC #include "audio_template.h" /* * Timer */ static int audio_is_timer_needed(AudioState *s) { HWVoiceIn *hwi = NULL; HWVoiceOut *hwo = NULL; while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) { if (!hwo->poll_mode) return 1; } while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) { if (!hwi->poll_mode) return 1; } return 0; } static void audio_reset_timer (AudioState *s) { if (audio_is_timer_needed(s)) { timer_mod_anticipate_ns(s->ts, qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks); if (!s->timer_running) { s->timer_running = true; s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); trace_audio_timer_start(s->period_ticks / SCALE_MS); } } else { timer_del(s->ts); if (s->timer_running) { s->timer_running = false; trace_audio_timer_stop(); } } } static void audio_timer (void *opaque) { int64_t now, diff; AudioState *s = opaque; now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); diff = now - s->timer_last; if (diff > s->period_ticks * 3 / 2) { trace_audio_timer_delayed(diff / SCALE_MS); } s->timer_last = now; audio_run(s, "timer"); audio_reset_timer(s); } /* * Public API */ size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size) { HWVoiceOut *hw; if (!sw) { /* XXX: Consider options */ return size; } hw = sw->hw; if (!hw->enabled) { dolog ("Writing to disabled voice %s\n", SW_NAME (sw)); return 0; } if (audio_get_pdo_out(hw->s->dev)->mixing_engine) { return audio_pcm_sw_write(sw, buf, size); } else { return hw->pcm_ops->write(hw, buf, size); } } size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size) { HWVoiceIn *hw; if (!sw) { /* XXX: Consider options */ return size; } hw = sw->hw; if (!hw->enabled) { dolog ("Reading from disabled voice %s\n", SW_NAME (sw)); return 0; } if (audio_get_pdo_in(hw->s->dev)->mixing_engine) { return audio_pcm_sw_read(sw, buf, size); } else { return hw->pcm_ops->read(hw, buf, size); } } int AUD_get_buffer_size_out(SWVoiceOut *sw) { return sw->hw->samples * sw->hw->info.bytes_per_frame; } void AUD_set_active_out (SWVoiceOut *sw, int on) { HWVoiceOut *hw; if (!sw) { return; } hw = sw->hw; if (sw->active != on) { AudioState *s = sw->s; SWVoiceOut *temp_sw; SWVoiceCap *sc; if (on) { hw->pending_disable = 0; if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { if (hw->pcm_ops->enable_out) { hw->pcm_ops->enable_out(hw, true); } 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 = sw->s; SWVoiceIn *temp_sw; if (on) { if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { if (hw->pcm_ops->enable_in) { hw->pcm_ops->enable_in(hw, true); } 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; if (hw->pcm_ops->enable_in) { hw->pcm_ops->enable_in(hw, false); } } } } sw->active = on; } } static size_t audio_get_avail (SWVoiceIn *sw) { size_t live; if (!sw) { return 0; } live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; if (audio_bug(__func__, live > sw->hw->conv_buf->size)) { dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live, sw->hw->conv_buf->size); return 0; } ldebug ( "%s: get_avail live %d ret %" PRId64 "\n", SW_NAME (sw), live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame ); return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame; } static size_t audio_get_free(SWVoiceOut *sw) { size_t live, dead; if (!sw) { return 0; } live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > sw->hw->mix_buf->size)) { dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live, sw->hw->mix_buf->size); return 0; } dead = sw->hw->mix_buf->size - 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.bytes_per_frame); #endif return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame; } static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, size_t samples) { size_t 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) { size_t till_end_of_hw = hw->mix_buf->size - rpos2; size_t to_write = MIN(till_end_of_hw, n); size_t bytes = to_write * hw->info.bytes_per_frame; size_t written; sw->buf = hw->mix_buf->samples + rpos2; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { dolog("Could not mix %zu bytes into a capture " "buffer, mixed %zu\n", bytes, written); break; } n -= to_write; rpos2 = (rpos2 + to_write) % hw->mix_buf->size; } } } n = MIN(samples, hw->mix_buf->size - rpos); mixeng_clear(hw->mix_buf->samples + rpos, n); mixeng_clear(hw->mix_buf->samples, samples - n); } static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) { size_t clipped = 0; while (live) { size_t size = live * hw->info.bytes_per_frame; size_t decr, proc; void *buf = hw->pcm_ops->get_buffer_out(hw, &size); if (size == 0) { break; } decr = MIN(size / hw->info.bytes_per_frame, live); if (buf) { audio_pcm_hw_clip_out(hw, buf, decr); } proc = hw->pcm_ops->put_buffer_out(hw, buf, decr * hw->info.bytes_per_frame) / hw->info.bytes_per_frame; live -= proc; clipped += proc; hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size; if (proc == 0 || proc < decr) { break; } } if (hw->pcm_ops->run_buffer_out) { hw->pcm_ops->run_buffer_out(hw); } return clipped; } static void audio_run_out (AudioState *s) { HWVoiceOut *hw = NULL; SWVoiceOut *sw; if (!audio_get_pdo_out(s->dev)->mixing_engine) { while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { /* there is exactly 1 sw for each hw with no mixeng */ sw = hw->sw_head.lh_first; if (hw->pending_disable) { hw->enabled = 0; hw->pending_disable = 0; if (hw->pcm_ops->enable_out) { hw->pcm_ops->enable_out(hw, false); } } if (sw->active) { sw->callback.fn(sw->callback.opaque, INT_MAX); } } return; } while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { size_t played, live, prev_rpos, free; int nb_live, cleanup_required; live = audio_pcm_hw_get_live_out (hw, &nb_live); if (!nb_live) { live = 0; } if (audio_bug(__func__, live > hw->mix_buf->size)) { dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); continue; } if (hw->pending_disable && !nb_live) { SWVoiceCap *sc; #ifdef DEBUG_OUT dolog ("Disabling voice\n"); #endif hw->enabled = 0; hw->pending_disable = 0; if (hw->pcm_ops->enable_out) { hw->pcm_ops->enable_out(hw, false); } 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); } } } if (hw->pcm_ops->run_buffer_out) { hw->pcm_ops->run_buffer_out(hw); } continue; } prev_rpos = hw->mix_buf->pos; played = audio_pcm_hw_run_out(hw, live); replay_audio_out(&played); if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) { dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n", hw->mix_buf->pos, hw->mix_buf->size, played); hw->mix_buf->pos = 0; } #ifdef DEBUG_OUT dolog("played=%zu\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=%zu sw->total_hw_samples_mixed=%zu\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 size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples) { size_t conv = 0; STSampleBuffer *conv_buf = hw->conv_buf; while (samples) { size_t proc; size_t size = samples * hw->info.bytes_per_frame; void *buf = hw->pcm_ops->get_buffer_in(hw, &size); assert(size % hw->info.bytes_per_frame == 0); if (size == 0) { break; } proc = MIN(size / hw->info.bytes_per_frame, conv_buf->size - conv_buf->pos); hw->conv(conv_buf->samples + conv_buf->pos, buf, proc); conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size; samples -= proc; conv += proc; hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame); } return conv; } static void audio_run_in (AudioState *s) { HWVoiceIn *hw = NULL; if (!audio_get_pdo_in(s->dev)->mixing_engine) { while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) { /* there is exactly 1 sw for each hw with no mixeng */ SWVoiceIn *sw = hw->sw_head.lh_first; if (sw->active) { sw->callback.fn(sw->callback.opaque, INT_MAX); } } return; } while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) { SWVoiceIn *sw; size_t captured = 0, min; if (replay_mode != REPLAY_MODE_PLAY) { captured = audio_pcm_hw_run_in( hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw)); } replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos, hw->conv_buf->size); 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) { size_t 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) { size_t live, rpos, captured; HWVoiceOut *hw = &cap->hw; SWVoiceOut *sw; captured = live = audio_pcm_hw_get_live_out (hw, NULL); rpos = hw->mix_buf->pos; while (live) { size_t left = hw->mix_buf->size - rpos; size_t to_capture = MIN(live, left); struct st_sample *src; struct capture_callback *cb; src = hw->mix_buf->samples + 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.bytes_per_frame); } rpos = (rpos + to_capture) % hw->mix_buf->size; live -= to_capture; } hw->mix_buf->pos = 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=%zu sw->total_hw_samples_mixed=%zu\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(AudioState *s, const char *msg) { 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 } void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size) { ssize_t start; if (unlikely(!hw->buf_emul)) { size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame; hw->buf_emul = g_malloc(calc_size); hw->size_emul = calc_size; hw->pos_emul = hw->pending_emul = 0; } while (hw->pending_emul < hw->size_emul) { size_t read_len = MIN(hw->size_emul - hw->pos_emul, hw->size_emul - hw->pending_emul); size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul, read_len); hw->pending_emul += read; hw->pos_emul = (hw->pos_emul + read) % hw->size_emul; if (read < read_len) { break; } } start = ((ssize_t) hw->pos_emul) - hw->pending_emul; if (start < 0) { start += hw->size_emul; } assert(start >= 0 && start < hw->size_emul); *size = MIN(*size, hw->pending_emul); *size = MIN(*size, hw->size_emul - start); return hw->buf_emul + start; } void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size) { assert(size <= hw->pending_emul); hw->pending_emul -= size; } void audio_generic_run_buffer_out(HWVoiceOut *hw) { while (hw->pending_emul) { size_t write_len, written; ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul; if (start < 0) { start += hw->size_emul; } assert(start >= 0 && start < hw->size_emul); write_len = MIN(hw->pending_emul, hw->size_emul - start); written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len); hw->pending_emul -= written; if (written < write_len) { break; } } } void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size) { if (unlikely(!hw->buf_emul)) { size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame; hw->buf_emul = g_malloc(calc_size); hw->size_emul = calc_size; hw->pos_emul = hw->pending_emul = 0; } *size = MIN(hw->size_emul - hw->pending_emul, hw->size_emul - hw->pos_emul); return hw->buf_emul + hw->pos_emul; } size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size) { assert(buf == hw->buf_emul + hw->pos_emul && size + hw->pending_emul <= hw->size_emul); hw->pending_emul += size; hw->pos_emul = (hw->pos_emul + size) % hw->size_emul; return size; } size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size) { size_t total = 0; while (total < size) { size_t dst_size = size - total; size_t copy_size, proc; void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size); if (dst_size == 0) { break; } copy_size = MIN(size - total, dst_size); if (dst) { memcpy(dst, (char *)buf + total, copy_size); } proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size); total += proc; if (proc == 0 || proc < copy_size) { break; } } if (hw->pcm_ops->run_buffer_out) { hw->pcm_ops->run_buffer_out(hw); } return total; } size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size) { size_t total = 0; while (total < size) { size_t src_size = size - total; void *src = hw->pcm_ops->get_buffer_in(hw, &src_size); if (src_size == 0) { break; } memcpy((char *)buf + total, src, src_size); hw->pcm_ops->put_buffer_in(hw, src, src_size); total += src_size; } return total; } static int audio_driver_init(AudioState *s, struct audio_driver *drv, bool msg, Audiodev *dev) { s->drv_opaque = drv->init(dev); if (s->drv_opaque) { if (!drv->pcm_ops->get_buffer_in) { drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in; drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in; } if (!drv->pcm_ops->get_buffer_out) { drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out; drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out; } audio_init_nb_voices_out(s, drv); audio_init_nb_voices_in(s, drv); s->drv = drv; return 0; } else { if (msg) { 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; s->vm_running = running; while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) { if (hwo->pcm_ops->enable_out) { hwo->pcm_ops->enable_out(hwo, running); } } while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) { if (hwi->pcm_ops->enable_in) { hwi->pcm_ops->enable_in(hwi, running); } } audio_reset_timer (s); } static void free_audio_state(AudioState *s) { HWVoiceOut *hwo, *hwon; HWVoiceIn *hwi, *hwin; QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) { SWVoiceCap *sc; if (hwo->enabled && hwo->pcm_ops->enable_out) { hwo->pcm_ops->enable_out(hwo, false); } 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, &s->hw_head_in, entries, hwin) { if (hwi->enabled && hwi->pcm_ops->enable_in) { hwi->pcm_ops->enable_in(hwi, false); } hwi->pcm_ops->fini_in (hwi); QLIST_REMOVE(hwi, entries); } if (s->drv) { s->drv->fini (s->drv_opaque); s->drv = NULL; } if (s->dev) { qapi_free_Audiodev(s->dev); s->dev = NULL; } if (s->ts) { timer_free(s->ts); s->ts = NULL; } g_free(s); } void audio_cleanup(void) { while (!QTAILQ_EMPTY(&audio_states)) { AudioState *s = QTAILQ_FIRST(&audio_states); QTAILQ_REMOVE(&audio_states, s, list); free_audio_state(s); } } static const VMStateDescription vmstate_audio = { .name = "audio", .version_id = 1, .minimum_version_id = 1, .fields = (VMStateField[]) { VMSTATE_END_OF_LIST() } }; static void audio_validate_opts(Audiodev *dev, Error **errp); static AudiodevListEntry *audiodev_find( AudiodevListHead *head, const char *drvname) { AudiodevListEntry *e; QSIMPLEQ_FOREACH(e, head, next) { if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) { return e; } } return NULL; } /* * if we have dev, this function was called because of an -audiodev argument => * initialize a new state with it * if dev == NULL => legacy implicit initialization, return the already created * state or create a new one */ static AudioState *audio_init(Audiodev *dev, const char *name) { static bool atexit_registered; size_t i; int done = 0; const char *drvname = NULL; VMChangeStateEntry *e; AudioState *s; struct audio_driver *driver; /* silence gcc warning about uninitialized variable */ AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head); if (using_spice) { /* * When using spice allow the spice audio driver being picked * as default. * * Temporary hack. Using audio devices without explicit * audiodev= property is already deprecated. Same goes for * the -soundhw switch. Once this support gets finally * removed we can also drop the concept of a default audio * backend and this can go away. */ driver = audio_driver_lookup("spice"); if (driver) { driver->can_be_default = 1; } } if (dev) { /* -audiodev option */ legacy_config = false; drvname = AudiodevDriver_str(dev->driver); } else if (!QTAILQ_EMPTY(&audio_states)) { if (!legacy_config) { dolog("Device %s: audiodev default parameter is deprecated, please " "specify audiodev=%s\n", name, QTAILQ_FIRST(&audio_states)->dev->id); } return QTAILQ_FIRST(&audio_states); } else { /* legacy implicit initialization */ head = audio_handle_legacy_opts(); /* * In case of legacy initialization, all Audiodevs in the list will have * the same configuration (except the driver), so it doesn't matter which * one we chose. We need an Audiodev to set up AudioState before we can * init a driver. Also note that dev at this point is still in the * list. */ dev = QSIMPLEQ_FIRST(&head)->dev; audio_validate_opts(dev, &error_abort); } s = g_malloc0(sizeof(AudioState)); s->dev = dev; QLIST_INIT (&s->hw_head_out); QLIST_INIT (&s->hw_head_in); QLIST_INIT (&s->cap_head); if (!atexit_registered) { atexit(audio_cleanup); atexit_registered = true; } QTAILQ_INSERT_TAIL(&audio_states, s, list); s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s); s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices; s->nb_hw_voices_in = audio_get_pdo_in(dev)->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; } if (drvname) { driver = audio_driver_lookup(drvname); if (driver) { done = !audio_driver_init(s, driver, true, dev); } else { dolog ("Unknown audio driver `%s'\n", drvname); } } else { for (i = 0; audio_prio_list[i]; i++) { AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]); driver = audio_driver_lookup(audio_prio_list[i]); if (e && driver) { s->dev = dev = e->dev; audio_validate_opts(dev, &error_abort); done = !audio_driver_init(s, driver, false, dev); if (done) { e->dev = NULL; break; } } } } audio_free_audiodev_list(&head); if (!done) { driver = audio_driver_lookup("none"); done = !audio_driver_init(s, driver, false, dev); assert(done); dolog("warning: Using timer based audio emulation\n"); } if (dev->timer_period <= 0) { s->period_ticks = 1; } else { s->period_ticks = dev->timer_period * (int64_t)SCALE_US; } 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); return s; } void audio_free_audiodev_list(AudiodevListHead *head) { AudiodevListEntry *e; while ((e = QSIMPLEQ_FIRST(head))) { QSIMPLEQ_REMOVE_HEAD(head, next); qapi_free_Audiodev(e->dev); g_free(e); } } void AUD_register_card (const char *name, QEMUSoundCard *card) { if (!card->state) { card->state = audio_init(NULL, name); } card->name = g_strdup (name); memset (&card->entries, 0, sizeof (card->entries)); QLIST_INSERT_HEAD(&card->state->card_head, card, entries); } void AUD_remove_card (QEMUSoundCard *card) { QLIST_REMOVE (card, entries); g_free (card->name); } CaptureVoiceOut *AUD_add_capture( AudioState *s, struct audsettings *as, struct audio_capture_ops *ops, void *cb_opaque ) { CaptureVoiceOut *cap; struct capture_callback *cb; if (!s) { if (!legacy_config) { dolog("Capturing without setting an audiodev is deprecated\n"); } s = audio_init(NULL, NULL); } if (!audio_get_pdo_out(s->dev)->mixing_engine) { dolog("Can't capture with mixeng disabled\n"); return NULL; } if (audio_validate_settings (as)) { dolog ("Invalid settings were passed when trying to add capture\n"); audio_print_settings (as); return NULL; } cb = g_malloc0(sizeof(*cb)); cb->ops = *ops; cb->opaque = cb_opaque; cap = audio_pcm_capture_find_specific(s, as); if (cap) { QLIST_INSERT_HEAD (&cap->cb_head, cb, entries); return cap; } else { HWVoiceOut *hw; CaptureVoiceOut *cap; cap = g_malloc0(sizeof(*cap)); hw = &cap->hw; hw->s = s; QLIST_INIT (&hw->sw_head); QLIST_INIT (&cap->cb_head); /* XXX find a more elegant way */ hw->samples = 4096 * 4; audio_pcm_hw_alloc_resources_out(hw); audio_pcm_init_info (&hw->info, as); cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); if (hw->info.is_float) { hw->clip = mixeng_clip_float[hw->info.nchannels == 2]; } else { hw->clip = mixeng_clip [hw->info.nchannels == 2] [hw->info.is_signed] [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, &s->hw_head_out, entries) { audio_attach_capture (hw); } return cap; } } 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) { Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } }; audio_set_volume_out(sw, &vol); } void audio_set_volume_out(SWVoiceOut *sw, Volume *vol) { if (sw) { HWVoiceOut *hw = sw->hw; sw->vol.mute = vol->mute; sw->vol.l = nominal_volume.l * vol->vol[0] / 255; sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] / 255; if (hw->pcm_ops->volume_out) { hw->pcm_ops->volume_out(hw, vol); } } } void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol) { Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } }; audio_set_volume_in(sw, &vol); } void audio_set_volume_in(SWVoiceIn *sw, Volume *vol) { if (sw) { HWVoiceIn *hw = sw->hw; sw->vol.mute = vol->mute; sw->vol.l = nominal_volume.l * vol->vol[0] / 255; sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] / 255; if (hw->pcm_ops->volume_in) { hw->pcm_ops->volume_in(hw, vol); } } } void audio_create_pdos(Audiodev *dev) { switch (dev->driver) { #define CASE(DRIVER, driver, pdo_name) \ case AUDIODEV_DRIVER_##DRIVER: \ if (!dev->u.driver.has_in) { \ dev->u.driver.in = g_malloc0( \ sizeof(Audiodev##pdo_name##PerDirectionOptions)); \ dev->u.driver.has_in = true; \ } \ if (!dev->u.driver.has_out) { \ dev->u.driver.out = g_malloc0( \ sizeof(Audiodev##pdo_name##PerDirectionOptions)); \ dev->u.driver.has_out = true; \ } \ break CASE(NONE, none, ); CASE(ALSA, alsa, Alsa); CASE(COREAUDIO, coreaudio, Coreaudio); CASE(DSOUND, dsound, ); CASE(JACK, jack, Jack); CASE(OSS, oss, Oss); CASE(PA, pa, Pa); CASE(SDL, sdl, Sdl); CASE(SPICE, spice, ); CASE(WAV, wav, ); case AUDIODEV_DRIVER__MAX: abort(); }; } static void audio_validate_per_direction_opts( AudiodevPerDirectionOptions *pdo, Error **errp) { if (!pdo->has_mixing_engine) { pdo->has_mixing_engine = true; pdo->mixing_engine = true; } if (!pdo->has_fixed_settings) { pdo->has_fixed_settings = true; pdo->fixed_settings = pdo->mixing_engine; } if (!pdo->fixed_settings && (pdo->has_frequency || pdo->has_channels || pdo->has_format)) { error_setg(errp, "You can't use frequency, channels or format with fixed-settings=off"); return; } if (!pdo->mixing_engine && pdo->fixed_settings) { error_setg(errp, "You can't use fixed-settings without mixeng"); return; } if (!pdo->has_frequency) { pdo->has_frequency = true; pdo->frequency = 44100; } if (!pdo->has_channels) { pdo->has_channels = true; pdo->channels = 2; } if (!pdo->has_voices) { pdo->has_voices = true; pdo->voices = pdo->mixing_engine ? 1 : INT_MAX; } if (!pdo->has_format) { pdo->has_format = true; pdo->format = AUDIO_FORMAT_S16; } } static void audio_validate_opts(Audiodev *dev, Error **errp) { Error *err = NULL; audio_create_pdos(dev); audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err); if (err) { error_propagate(errp, err); return; } audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err); if (err) { error_propagate(errp, err); return; } if (!dev->has_timer_period) { dev->has_timer_period = true; dev->timer_period = 10000; /* 100Hz -> 10ms */ } } void audio_parse_option(const char *opt) { AudiodevListEntry *e; Audiodev *dev = NULL; Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal); visit_type_Audiodev(v, NULL, &dev, &error_fatal); visit_free(v); audio_validate_opts(dev, &error_fatal); e = g_malloc0(sizeof(AudiodevListEntry)); e->dev = dev; QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next); } void audio_init_audiodevs(void) { AudiodevListEntry *e; QSIMPLEQ_FOREACH(e, &audiodevs, next) { audio_init(e->dev, NULL); } } audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo) { return (audsettings) { .freq = pdo->frequency, .nchannels = pdo->channels, .fmt = pdo->format, .endianness = AUDIO_HOST_ENDIANNESS, }; } int audioformat_bytes_per_sample(AudioFormat fmt) { switch (fmt) { case AUDIO_FORMAT_U8: case AUDIO_FORMAT_S8: return 1; case AUDIO_FORMAT_U16: case AUDIO_FORMAT_S16: return 2; case AUDIO_FORMAT_U32: case AUDIO_FORMAT_S32: case AUDIO_FORMAT_F32: return 4; case AUDIO_FORMAT__MAX: ; } abort(); } /* frames = freq * usec / 1e6 */ int audio_buffer_frames(AudiodevPerDirectionOptions *pdo, audsettings *as, int def_usecs) { uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs; return (as->freq * usecs + 500000) / 1000000; } /* samples = channels * frames = channels * freq * usec / 1e6 */ int audio_buffer_samples(AudiodevPerDirectionOptions *pdo, audsettings *as, int def_usecs) { return as->nchannels * audio_buffer_frames(pdo, as, def_usecs); } /* * bytes = bytes_per_sample * samples = * bytes_per_sample * channels * freq * usec / 1e6 */ int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo, audsettings *as, int def_usecs) { return audio_buffer_samples(pdo, as, def_usecs) * audioformat_bytes_per_sample(as->fmt); } AudioState *audio_state_by_name(const char *name) { AudioState *s; QTAILQ_FOREACH(s, &audio_states, list) { assert(s->dev); if (strcmp(name, s->dev->id) == 0) { return s; } } return NULL; } const char *audio_get_id(QEMUSoundCard *card) { if (card->state) { assert(card->state->dev); return card->state->dev->id; } else { return ""; } } void audio_rate_start(RateCtl *rate) { memset(rate, 0, sizeof(RateCtl)); rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); } size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate, size_t bytes_avail) { int64_t now; int64_t ticks; int64_t bytes; int64_t samples; size_t ret; now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); ticks = now - rate->start_ticks; bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND); samples = (bytes - rate->bytes_sent) / info->bytes_per_frame; if (samples < 0 || samples > 65536) { AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples); audio_rate_start(rate); samples = 0; } ret = MIN(samples * info->bytes_per_frame, bytes_avail); rate->bytes_sent += ret; return ret; }