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ed2a4a7941
This adds proper support for float samples in mixeng by adding a new audio format for it. Limitations: only native endianness is supported. None of the virtual sound cards support float samples (it looks like most of them only support 8 and 16 bit, only hda supports 32 bit), it is only used for the audio backends (i.e. host side). Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2184 lines
56 KiB
C
2184 lines
56 KiB
C
/*
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* QEMU Audio subsystem
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*
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* Copyright (c) 2003-2005 Vassili Karpov (malc)
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "qemu/osdep.h"
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#include "audio.h"
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#include "migration/vmstate.h"
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#include "monitor/monitor.h"
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#include "qemu/timer.h"
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#include "qapi/error.h"
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#include "qapi/qobject-input-visitor.h"
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#include "qapi/qapi-visit-audio.h"
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#include "qemu/cutils.h"
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#include "qemu/module.h"
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#include "sysemu/replay.h"
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#include "sysemu/runstate.h"
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#include "trace.h"
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#define AUDIO_CAP "audio"
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#include "audio_int.h"
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/* #define DEBUG_LIVE */
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/* #define DEBUG_OUT */
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/* #define DEBUG_CAPTURE */
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/* #define DEBUG_POLL */
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#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
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/* Order of CONFIG_AUDIO_DRIVERS is import.
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The 1st one is the one used by default, that is the reason
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that we generate the list.
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*/
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const char *audio_prio_list[] = {
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"spice",
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CONFIG_AUDIO_DRIVERS
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"none",
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"wav",
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NULL
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};
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static QLIST_HEAD(, audio_driver) audio_drivers;
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static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
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void audio_driver_register(audio_driver *drv)
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{
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QLIST_INSERT_HEAD(&audio_drivers, drv, next);
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}
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audio_driver *audio_driver_lookup(const char *name)
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{
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struct audio_driver *d;
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QLIST_FOREACH(d, &audio_drivers, next) {
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if (strcmp(name, d->name) == 0) {
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return d;
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}
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}
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audio_module_load_one(name);
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QLIST_FOREACH(d, &audio_drivers, next) {
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if (strcmp(name, d->name) == 0) {
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return d;
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}
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}
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return NULL;
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}
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static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
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QTAILQ_HEAD_INITIALIZER(audio_states);
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const struct mixeng_volume nominal_volume = {
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.mute = 0,
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#ifdef FLOAT_MIXENG
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.r = 1.0,
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.l = 1.0,
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#else
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.r = 1ULL << 32,
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.l = 1ULL << 32,
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#endif
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};
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static bool legacy_config = true;
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#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
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#error No its not
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#else
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int audio_bug (const char *funcname, int cond)
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{
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if (cond) {
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static int shown;
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AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
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if (!shown) {
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shown = 1;
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AUD_log (NULL, "Save all your work and restart without audio\n");
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AUD_log (NULL, "I am sorry\n");
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}
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AUD_log (NULL, "Context:\n");
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#if defined AUDIO_BREAKPOINT_ON_BUG
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# if defined HOST_I386
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# if defined __GNUC__
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__asm__ ("int3");
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# elif defined _MSC_VER
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_asm _emit 0xcc;
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# else
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abort ();
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# endif
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# else
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abort ();
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# endif
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#endif
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}
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return cond;
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}
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#endif
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static inline int audio_bits_to_index (int bits)
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{
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switch (bits) {
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case 8:
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return 0;
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case 16:
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return 1;
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case 32:
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return 2;
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default:
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audio_bug ("bits_to_index", 1);
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AUD_log (NULL, "invalid bits %d\n", bits);
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return 0;
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}
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}
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void *audio_calloc (const char *funcname, int nmemb, size_t size)
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{
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int cond;
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size_t len;
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len = nmemb * size;
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cond = !nmemb || !size;
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cond |= nmemb < 0;
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cond |= len < size;
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if (audio_bug ("audio_calloc", cond)) {
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AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
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funcname);
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AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
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return NULL;
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}
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return g_malloc0 (len);
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}
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void AUD_vlog (const char *cap, const char *fmt, va_list ap)
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{
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if (cap) {
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fprintf(stderr, "%s: ", cap);
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}
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vfprintf(stderr, fmt, ap);
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}
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void AUD_log (const char *cap, const char *fmt, ...)
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{
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va_list ap;
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va_start (ap, fmt);
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AUD_vlog (cap, fmt, ap);
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va_end (ap);
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}
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static void audio_print_settings (struct audsettings *as)
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{
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dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
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switch (as->fmt) {
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case AUDIO_FORMAT_S8:
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AUD_log (NULL, "S8");
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break;
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case AUDIO_FORMAT_U8:
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AUD_log (NULL, "U8");
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break;
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case AUDIO_FORMAT_S16:
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AUD_log (NULL, "S16");
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break;
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case AUDIO_FORMAT_U16:
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AUD_log (NULL, "U16");
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break;
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case AUDIO_FORMAT_S32:
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AUD_log (NULL, "S32");
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break;
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case AUDIO_FORMAT_U32:
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AUD_log (NULL, "U32");
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break;
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case AUDIO_FORMAT_F32:
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AUD_log (NULL, "F32");
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break;
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default:
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AUD_log (NULL, "invalid(%d)", as->fmt);
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break;
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}
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AUD_log (NULL, " endianness=");
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switch (as->endianness) {
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case 0:
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AUD_log (NULL, "little");
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break;
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case 1:
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AUD_log (NULL, "big");
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break;
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default:
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AUD_log (NULL, "invalid");
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break;
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}
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AUD_log (NULL, "\n");
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}
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static int audio_validate_settings (struct audsettings *as)
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{
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int invalid;
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invalid = as->nchannels < 1;
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invalid |= as->endianness != 0 && as->endianness != 1;
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switch (as->fmt) {
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case AUDIO_FORMAT_S8:
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case AUDIO_FORMAT_U8:
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case AUDIO_FORMAT_S16:
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case AUDIO_FORMAT_U16:
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case AUDIO_FORMAT_S32:
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case AUDIO_FORMAT_U32:
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case AUDIO_FORMAT_F32:
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break;
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default:
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invalid = 1;
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break;
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}
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invalid |= as->freq <= 0;
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return invalid ? -1 : 0;
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}
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static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
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{
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int bits = 8;
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bool is_signed = false, is_float = false;
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switch (as->fmt) {
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case AUDIO_FORMAT_S8:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U8:
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break;
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case AUDIO_FORMAT_S16:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U16:
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bits = 16;
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break;
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case AUDIO_FORMAT_F32:
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is_float = true;
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/* fall through */
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case AUDIO_FORMAT_S32:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U32:
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bits = 32;
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break;
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default:
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abort();
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}
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return info->freq == as->freq
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&& info->nchannels == as->nchannels
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&& info->is_signed == is_signed
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&& info->is_float == is_float
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&& info->bits == bits
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&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
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}
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void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
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{
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int bits = 8, mul;
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bool is_signed = false, is_float = false;
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switch (as->fmt) {
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case AUDIO_FORMAT_S8:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U8:
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mul = 1;
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break;
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case AUDIO_FORMAT_S16:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U16:
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bits = 16;
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mul = 2;
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break;
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case AUDIO_FORMAT_F32:
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is_float = true;
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/* fall through */
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case AUDIO_FORMAT_S32:
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is_signed = true;
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/* fall through */
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case AUDIO_FORMAT_U32:
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bits = 32;
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mul = 4;
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break;
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default:
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abort();
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}
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info->freq = as->freq;
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info->bits = bits;
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info->is_signed = is_signed;
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info->is_float = is_float;
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info->nchannels = as->nchannels;
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info->bytes_per_frame = as->nchannels * mul;
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info->bytes_per_second = info->freq * info->bytes_per_frame;
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info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
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}
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void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
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{
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if (!len) {
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return;
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}
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if (info->is_signed || info->is_float) {
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memset(buf, 0x00, len * info->bytes_per_frame);
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}
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else {
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switch (info->bits) {
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case 8:
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memset(buf, 0x80, len * info->bytes_per_frame);
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break;
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case 16:
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{
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int i;
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uint16_t *p = buf;
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short s = INT16_MAX;
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if (info->swap_endianness) {
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s = bswap16 (s);
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}
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for (i = 0; i < len * info->nchannels; i++) {
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p[i] = s;
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}
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}
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break;
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case 32:
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{
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int i;
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uint32_t *p = buf;
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int32_t s = INT32_MAX;
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if (info->swap_endianness) {
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s = bswap32 (s);
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}
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for (i = 0; i < len * info->nchannels; i++) {
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p[i] = s;
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}
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}
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break;
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default:
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AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
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info->bits);
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break;
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}
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}
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}
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/*
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* Capture
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*/
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static void noop_conv (struct st_sample *dst, const void *src, int samples)
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{
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(void) src;
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(void) dst;
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(void) samples;
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}
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static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
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struct audsettings *as)
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{
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CaptureVoiceOut *cap;
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for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
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if (audio_pcm_info_eq (&cap->hw.info, as)) {
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return cap;
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}
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}
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return NULL;
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}
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static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
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{
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struct capture_callback *cb;
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#ifdef DEBUG_CAPTURE
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dolog ("notification %d sent\n", cmd);
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#endif
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for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
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cb->ops.notify (cb->opaque, cmd);
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}
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}
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static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
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{
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if (cap->hw.enabled != enabled) {
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audcnotification_e cmd;
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cap->hw.enabled = enabled;
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cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
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audio_notify_capture (cap, cmd);
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}
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}
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static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
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{
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HWVoiceOut *hw = &cap->hw;
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SWVoiceOut *sw;
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int enabled = 0;
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for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
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if (sw->active) {
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enabled = 1;
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break;
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}
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}
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audio_capture_maybe_changed (cap, enabled);
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}
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static void audio_detach_capture (HWVoiceOut *hw)
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{
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SWVoiceCap *sc = hw->cap_head.lh_first;
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while (sc) {
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SWVoiceCap *sc1 = sc->entries.le_next;
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SWVoiceOut *sw = &sc->sw;
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CaptureVoiceOut *cap = sc->cap;
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int was_active = sw->active;
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if (sw->rate) {
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st_rate_stop (sw->rate);
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sw->rate = NULL;
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}
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QLIST_REMOVE (sw, entries);
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QLIST_REMOVE (sc, entries);
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g_free (sc);
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if (was_active) {
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/* We have removed soft voice from the capture:
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this might have changed the overall status of the capture
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since this might have been the only active voice */
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audio_recalc_and_notify_capture (cap);
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}
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sc = sc1;
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}
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}
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static int audio_attach_capture (HWVoiceOut *hw)
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{
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AudioState *s = hw->s;
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CaptureVoiceOut *cap;
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audio_detach_capture (hw);
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for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
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SWVoiceCap *sc;
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SWVoiceOut *sw;
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HWVoiceOut *hw_cap = &cap->hw;
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sc = g_malloc0(sizeof(*sc));
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sc->cap = cap;
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sw = &sc->sw;
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sw->hw = hw_cap;
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sw->info = hw->info;
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sw->empty = 1;
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sw->active = hw->enabled;
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sw->conv = noop_conv;
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sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
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sw->vol = nominal_volume;
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sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
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if (!sw->rate) {
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dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
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g_free (sw);
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return -1;
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}
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QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
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QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
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#ifdef DEBUG_CAPTURE
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sw->name = g_strdup_printf ("for %p %d,%d,%d",
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hw, sw->info.freq, sw->info.bits,
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sw->info.nchannels);
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dolog ("Added %s active = %d\n", sw->name, sw->active);
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#endif
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if (sw->active) {
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audio_capture_maybe_changed (cap, 1);
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}
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}
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return 0;
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}
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|
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/*
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* Hard voice (capture)
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*/
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static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
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{
|
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SWVoiceIn *sw;
|
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size_t m = hw->total_samples_captured;
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|
|
|
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->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->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, decr, proc;
|
|
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
|
|
if (!buf || size == 0) {
|
|
break;
|
|
}
|
|
|
|
decr = MIN(size / hw->info.bytes_per_frame, live);
|
|
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);
|
|
}
|
|
}
|
|
}
|
|
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) {
|
|
hw->pcm_ops->put_buffer_in(hw, buf, size);
|
|
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 dst_size, copy_size;
|
|
void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
|
|
copy_size = MIN(size, dst_size);
|
|
|
|
memcpy(dst, buf, copy_size);
|
|
return hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
|
|
}
|
|
|
|
size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
|
|
{
|
|
size_t src_size, copy_size;
|
|
void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
|
|
copy_size = MIN(size, src_size);
|
|
|
|
memcpy(buf, src, copy_size);
|
|
hw->pcm_ops->put_buffer_in(hw, src, copy_size);
|
|
return copy_size;
|
|
}
|
|
|
|
|
|
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 bool is_cleaning_up;
|
|
|
|
bool audio_is_cleaning_up(void)
|
|
{
|
|
return is_cleaning_up;
|
|
}
|
|
|
|
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)
|
|
{
|
|
is_cleaning_up = true;
|
|
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 (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 does'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(OSS, oss, Oss);
|
|
CASE(PA, pa, Pa);
|
|
CASE(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;
|
|
}
|