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authorTimos Ampelikiotis <t.ampelikiotis@virtualopensystems.com>2023-10-10 11:40:56 +0000
committerTimos Ampelikiotis <t.ampelikiotis@virtualopensystems.com>2023-10-10 11:40:56 +0000
commite02cda008591317b1625707ff8e115a4841aa889 (patch)
treeaee302e3cf8b59ec2d32ec481be3d1afddfc8968 /audio/audio.c
parentcc668e6b7e0ffd8c9d130513d12053cf5eda1d3b (diff)
Introduce Virtio-loopback epsilon release:
Epsilon release introduces a new compatibility layer which make virtio-loopback design to work with QEMU and rust-vmm vhost-user backend without require any changes. Signed-off-by: Timos Ampelikiotis <t.ampelikiotis@virtualopensystems.com> Change-Id: I52e57563e08a7d0bdc002f8e928ee61ba0c53dd9
Diffstat (limited to 'audio/audio.c')
-rw-r--r--audio/audio.c2222
1 files changed, 2222 insertions, 0 deletions
diff --git a/audio/audio.c b/audio/audio.c
new file mode 100644
index 000000000..54a153c0e
--- /dev/null
+++ b/audio/audio.c
@@ -0,0 +1,2222 @@
+/*
+ * 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 "qemu-common.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 %zu\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 %zu 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 %zu dead %zu 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;
+
+ 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);
+ }
+
+ 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;
+ }
+
+ if (sw->active) {
+ free = audio_get_free (sw);
+ if (free > 0) {
+ sw->callback.fn (sw->callback.opaque, free);
+ }
+ }
+ }
+ }
+}
+
+static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
+{
+ size_t conv = 0;
+ STSampleBuffer *conv_buf = hw->conv_buf;
+
+ if (hw->pcm_ops->run_buffer_in) {
+ hw->pcm_ops->run_buffer_in(hw);
+ }
+
+ 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_run_buffer_in(HWVoiceIn *hw)
+{
+ if (unlikely(!hw->buf_emul)) {
+ hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+ hw->buf_emul = g_malloc(hw->size_emul);
+ 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;
+ }
+ }
+}
+
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+ 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);
+
+ *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)) {
+ hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+ hw->buf_emul = g_malloc(hw->size_emul);
+ 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;
+
+ if (hw->pcm_ops->run_buffer_in) {
+ hw->pcm_ops->run_buffer_in(hw);
+ }
+
+ 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, bool 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 bool vmstate_audio_needed(void *opaque)
+{
+ /*
+ * Never needed, this vmstate only exists in case
+ * an old qemu sends it to us.
+ */
+ return false;
+}
+
+static const VMStateDescription vmstate_audio = {
+ .name = "audio",
+ .version_id = 1,
+ .minimum_version_id = 1,
+ .needed = vmstate_audio_needed,
+ .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 "";
+ }
+}
+
+const char *audio_application_name(void)
+{
+ const char *vm_name;
+
+ vm_name = qemu_get_vm_name();
+ return vm_name ? vm_name : "qemu";
+}
+
+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;
+}