e1e6a6fcc9
Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de>
610 lines
15 KiB
C
610 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;
|
|
}
|
|
|
|
/*
|
|
* Allocate one additional audio frame that is needed for upsampling
|
|
* if the resample buffer size is small. For large buffer sizes take
|
|
* care of overflows.
|
|
*/
|
|
samples = samples < INT_MAX ? samples + 1 : INT_MAX;
|
|
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
|