qemu-e2k/audio/audio_template.h
Volker Rümelin 2c3f9a0a92 audio: change type and name of the resample buffer
Change the type of the resample buffer from struct st_sample *
to STSampleBuffer. Also change the name from buf to resample_buf
for better readability.

The new variables resample_buf.size and resample_buf.pos will be
used after the next patches. There is no functional change.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-2-vr_qemu@t-online.de>
2023-03-06 10:30:23 +04:00

604 lines
15 KiB
C

/*
* QEMU Audio subsystem header
*
* Copyright (c) 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.
*/
#ifdef DAC
#define NAME "playback"
#define HWBUF hw->mix_buf
#define TYPE out
#define HW HWVoiceOut
#define SW SWVoiceOut
#else
#define NAME "capture"
#define TYPE in
#define HW HWVoiceIn
#define SW SWVoiceIn
#define HWBUF hw->conv_buf
#endif
static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
struct audio_driver *drv)
{
int max_voices = glue (drv->max_voices_, TYPE);
size_t voice_size = glue(drv->voice_size_, TYPE);
if (glue (s->nb_hw_voices_, TYPE) > max_voices) {
if (!max_voices) {
#ifdef DAC
dolog ("Driver `%s' does not support " NAME "\n", drv->name);
#endif
} else {
dolog ("Driver `%s' does not support %d " NAME " voices, max %d\n",
drv->name,
glue (s->nb_hw_voices_, TYPE),
max_voices);
}
glue (s->nb_hw_voices_, TYPE) = max_voices;
}
if (audio_bug(__func__, !voice_size && max_voices)) {
dolog ("drv=`%s' voice_size=0 max_voices=%d\n",
drv->name, max_voices);
glue (s->nb_hw_voices_, TYPE) = 0;
}
if (audio_bug(__func__, voice_size && !max_voices)) {
dolog("drv=`%s' voice_size=%zu max_voices=0\n",
drv->name, voice_size);
}
}
static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
{
g_free(hw->buf_emul);
g_free(HWBUF.buffer);
HWBUF.buffer = NULL;
HWBUF.size = 0;
}
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
if (glue(audio_get_pdo_, TYPE)(hw->s->dev)->mixing_engine) {
size_t samples = hw->samples;
if (audio_bug(__func__, samples == 0)) {
dolog("Attempted to allocate empty buffer\n");
}
HWBUF.buffer = g_new0(st_sample, samples);
HWBUF.size = samples;
HWBUF.pos = 0;
} else {
HWBUF.buffer = NULL;
HWBUF.size = 0;
}
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
{
g_free(sw->resample_buf.buffer);
sw->resample_buf.buffer = NULL;
sw->resample_buf.size = 0;
if (sw->rate) {
st_rate_stop (sw->rate);
}
sw->rate = NULL;
}
static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
return 0;
}
#ifdef DAC
samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
#else
samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
#endif
if (audio_bug(__func__, samples < 0)) {
dolog("Can not allocate buffer for `%s' (%d samples)\n",
SW_NAME(sw), samples);
return -1;
}
if (samples == 0) {
HW *hw = sw->hw;
size_t f_fe_min;
/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size;
qemu_log_mask(LOG_UNIMP,
AUDIO_CAP ": The guest selected a " NAME " sample rate"
" of %d Hz for %s. Only sample rates >= %zu Hz are"
" supported.\n",
sw->info.freq, sw->name, f_fe_min);
return -1;
}
sw->resample_buf.buffer = g_new0(st_sample, samples);
sw->resample_buf.size = samples;
sw->resample_buf.pos = 0;
#ifdef DAC
sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
#else
sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
#endif
return 0;
}
static int glue (audio_pcm_sw_init_, TYPE) (
SW *sw,
HW *hw,
const char *name,
struct audsettings *as
)
{
int err;
audio_pcm_init_info (&sw->info, as);
sw->hw = hw;
sw->active = 0;
#ifdef DAC
sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
sw->total_hw_samples_mixed = 0;
sw->empty = 1;
#else
sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
#endif
if (sw->info.is_float) {
#ifdef DAC
sw->conv = mixeng_conv_float[sw->info.nchannels == 2];
#else
sw->clip = mixeng_clip_float[sw->info.nchannels == 2];
#endif
} else {
#ifdef DAC
sw->conv = mixeng_conv
#else
sw->clip = mixeng_clip
#endif
[sw->info.nchannels == 2]
[sw->info.is_signed]
[sw->info.swap_endianness]
[audio_bits_to_index(sw->info.bits)];
}
sw->name = g_strdup (name);
err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
if (err) {
g_free (sw->name);
sw->name = NULL;
}
return err;
}
static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
{
glue (audio_pcm_sw_free_resources_, TYPE) (sw);
g_free (sw->name);
sw->name = NULL;
}
static void glue (audio_pcm_hw_add_sw_, TYPE) (HW *hw, SW *sw)
{
QLIST_INSERT_HEAD (&hw->sw_head, sw, entries);
}
static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
{
QLIST_REMOVE (sw, entries);
}
static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
{
HW *hw = *hwp;
AudioState *s = hw->s;
if (!hw->sw_head.lh_first) {
#ifdef DAC
audio_detach_capture(hw);
#endif
QLIST_REMOVE(hw, entries);
glue(hw->pcm_ops->fini_, TYPE) (hw);
glue(s->nb_hw_voices_, TYPE) += 1;
glue(audio_pcm_hw_free_resources_ , TYPE) (hw);
g_free(hw);
*hwp = NULL;
}
}
static HW *glue(audio_pcm_hw_find_any_, TYPE)(AudioState *s, HW *hw)
{
return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first;
}
static HW *glue(audio_pcm_hw_find_any_enabled_, TYPE)(AudioState *s, HW *hw)
{
while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (hw->enabled) {
return hw;
}
}
return NULL;
}
static HW *glue(audio_pcm_hw_find_specific_, TYPE)(AudioState *s, HW *hw,
struct audsettings *as)
{
while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (audio_pcm_info_eq (&hw->info, as)) {
return hw;
}
}
return NULL;
}
static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
struct audsettings *as)
{
HW *hw;
struct audio_driver *drv = s->drv;
if (!glue (s->nb_hw_voices_, TYPE)) {
return NULL;
}
if (audio_bug(__func__, !drv)) {
dolog ("No host audio driver\n");
return NULL;
}
if (audio_bug(__func__, !drv->pcm_ops)) {
dolog ("Host audio driver without pcm_ops\n");
return NULL;
}
/*
* Since glue(s->nb_hw_voices_, TYPE) is != 0, glue(drv->voice_size_, TYPE)
* is guaranteed to be != 0. See the audio_init_nb_voices_* functions.
*/
hw = g_malloc0(glue(drv->voice_size_, TYPE));
hw->s = s;
hw->pcm_ops = drv->pcm_ops;
QLIST_INIT (&hw->sw_head);
#ifdef DAC
QLIST_INIT (&hw->cap_head);
#endif
if (glue (hw->pcm_ops->init_, TYPE) (hw, as, s->drv_opaque)) {
goto err0;
}
if (audio_bug(__func__, hw->samples <= 0)) {
dolog("hw->samples=%zd\n", hw->samples);
goto err1;
}
if (hw->info.is_float) {
#ifdef DAC
hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
#else
hw->conv = mixeng_conv_float[hw->info.nchannels == 2];
#endif
} else {
#ifdef DAC
hw->clip = mixeng_clip
#else
hw->conv = mixeng_conv
#endif
[hw->info.nchannels == 2]
[hw->info.is_signed]
[hw->info.swap_endianness]
[audio_bits_to_index(hw->info.bits)];
}
glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
QLIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
glue (s->nb_hw_voices_, TYPE) -= 1;
#ifdef DAC
audio_attach_capture (hw);
#endif
return hw;
err1:
glue (hw->pcm_ops->fini_, TYPE) (hw);
err0:
g_free (hw);
return NULL;
}
AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
{
switch (dev->driver) {
case AUDIODEV_DRIVER_NONE:
return dev->u.none.TYPE;
#ifdef CONFIG_AUDIO_ALSA
case AUDIODEV_DRIVER_ALSA:
return qapi_AudiodevAlsaPerDirectionOptions_base(dev->u.alsa.TYPE);
#endif
#ifdef CONFIG_AUDIO_COREAUDIO
case AUDIODEV_DRIVER_COREAUDIO:
return qapi_AudiodevCoreaudioPerDirectionOptions_base(
dev->u.coreaudio.TYPE);
#endif
#ifdef CONFIG_DBUS_DISPLAY
case AUDIODEV_DRIVER_DBUS:
return dev->u.dbus.TYPE;
#endif
#ifdef CONFIG_AUDIO_DSOUND
case AUDIODEV_DRIVER_DSOUND:
return dev->u.dsound.TYPE;
#endif
#ifdef CONFIG_AUDIO_JACK
case AUDIODEV_DRIVER_JACK:
return qapi_AudiodevJackPerDirectionOptions_base(dev->u.jack.TYPE);
#endif
#ifdef CONFIG_AUDIO_OSS
case AUDIODEV_DRIVER_OSS:
return qapi_AudiodevOssPerDirectionOptions_base(dev->u.oss.TYPE);
#endif
#ifdef CONFIG_AUDIO_PA
case AUDIODEV_DRIVER_PA:
return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
#endif
#ifdef CONFIG_AUDIO_SDL
case AUDIODEV_DRIVER_SDL:
return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE);
#endif
#ifdef CONFIG_AUDIO_SNDIO
case AUDIODEV_DRIVER_SNDIO:
return dev->u.sndio.TYPE;
#endif
#ifdef CONFIG_SPICE
case AUDIODEV_DRIVER_SPICE:
return dev->u.spice.TYPE;
#endif
case AUDIODEV_DRIVER_WAV:
return dev->u.wav.TYPE;
case AUDIODEV_DRIVER__MAX:
break;
}
abort();
}
static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
{
HW *hw;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (!pdo->mixing_engine || pdo->fixed_settings) {
hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (!pdo->mixing_engine || hw) {
return hw;
}
}
hw = glue(audio_pcm_hw_find_specific_, TYPE)(s, NULL, as);
if (hw) {
return hw;
}
hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (hw) {
return hw;
}
return glue(audio_pcm_hw_find_any_, TYPE)(s, NULL);
}
static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
AudioState *s,
const char *sw_name,
struct audsettings *as
)
{
SW *sw;
HW *hw;
struct audsettings hw_as;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (pdo->fixed_settings) {
hw_as = audiodev_to_audsettings(pdo);
} else {
hw_as = *as;
}
sw = g_new0(SW, 1);
sw->s = s;
hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
if (!hw) {
dolog("Could not create a backend for voice `%s'\n", sw_name);
goto err1;
}
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
goto err2;
}
return sw;
err2:
glue (audio_pcm_hw_del_sw_, TYPE) (sw);
glue (audio_pcm_hw_gc_, TYPE) (&hw);
err1:
g_free(sw);
return NULL;
}
static void glue (audio_close_, TYPE) (SW *sw)
{
glue (audio_pcm_sw_fini_, TYPE) (sw);
glue (audio_pcm_hw_del_sw_, TYPE) (sw);
glue (audio_pcm_hw_gc_, TYPE) (&sw->hw);
g_free (sw);
}
void glue (AUD_close_, TYPE) (QEMUSoundCard *card, SW *sw)
{
if (sw) {
if (audio_bug(__func__, !card)) {
dolog ("card=%p\n", card);
return;
}
glue (audio_close_, TYPE) (sw);
}
}
SW *glue (AUD_open_, TYPE) (
QEMUSoundCard *card,
SW *sw,
const char *name,
void *callback_opaque ,
audio_callback_fn callback_fn,
struct audsettings *as
)
{
AudioState *s;
AudiodevPerDirectionOptions *pdo;
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
card, name, callback_fn, as);
goto fail;
}
s = card->state;
pdo = glue(audio_get_pdo_, TYPE)(s->dev);
ldebug ("open %s, freq %d, nchannels %d, fmt %d\n",
name, as->freq, as->nchannels, as->fmt);
if (audio_bug(__func__, audio_validate_settings(as))) {
audio_print_settings (as);
goto fail;
}
if (audio_bug(__func__, !s->drv)) {
dolog ("Can not open `%s' (no host audio driver)\n", name);
goto fail;
}
if (sw && audio_pcm_info_eq (&sw->info, as)) {
return sw;
}
if (!pdo->fixed_settings && sw) {
glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
if (sw) {
HW *hw = sw->hw;
if (!hw) {
dolog("Internal logic error: voice `%s' has no backend\n",
SW_NAME(sw));
goto fail;
}
glue (audio_pcm_sw_fini_, TYPE) (sw);
if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
goto fail;
}
} else {
sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
if (!sw) {
return NULL;
}
}
sw->card = card;
sw->vol = nominal_volume;
sw->callback.fn = callback_fn;
sw->callback.opaque = callback_opaque;
#ifdef DEBUG_AUDIO
dolog ("%s\n", name);
audio_pcm_print_info ("hw", &sw->hw->info);
audio_pcm_print_info ("sw", &sw->info);
#endif
return sw;
fail:
glue (AUD_close_, TYPE) (card, sw);
return NULL;
}
int glue (AUD_is_active_, TYPE) (SW *sw)
{
return sw ? sw->active : 0;
}
void glue (AUD_init_time_stamp_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
{
if (!sw) {
return;
}
ts->old_ts = sw->hw->ts_helper;
}
uint64_t glue (AUD_get_elapsed_usec_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
{
uint64_t delta, cur_ts, old_ts;
if (!sw) {
return 0;
}
cur_ts = sw->hw->ts_helper;
old_ts = ts->old_ts;
/* dolog ("cur %" PRId64 " old %" PRId64 "\n", cur_ts, old_ts); */
if (cur_ts >= old_ts) {
delta = cur_ts - old_ts;
} else {
delta = UINT64_MAX - old_ts + cur_ts;
}
if (!delta) {
return 0;
}
return muldiv64 (delta, sw->hw->info.freq, 1000000);
}
#undef TYPE
#undef HW
#undef SW
#undef HWBUF
#undef NAME