qemu-e2k/audio/audio_template.h

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

609 lines
15 KiB
C
Raw Permalink Normal View History

/*
* 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 min_voices)
{
int max_voices = glue (drv->max_voices_, TYPE);
size_t voice_size = glue(drv->voice_size_, TYPE);
glue (s->nb_hw_voices_, TYPE) = glue(audio_get_pdo_, TYPE)(s->dev)->voices;
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 (glue (s->nb_hw_voices_, TYPE) < min_voices) {
dolog ("Bogus number of " NAME " voices %d, setting to %d\n",
glue (s->nb_hw_voices_, TYPE),
min_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)
{
HW *hw = sw->hw;
uint64_t samples;
if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
return 0;
}
samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
audio: log unimplemented audio device sample rates Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-01-21 10:47:25 +01:00
if (samples == 0) {
uint64_t f_fe_min;
uint64_t f_be = (uint32_t)hw->info.freq;
audio: log unimplemented audio device sample rates Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-01-21 10:47:25 +01:00
/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
audio: log unimplemented audio device sample rates Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-01-21 10:47:25 +01:00
qemu_log_mask(LOG_UNIMP,
AUDIO_CAP ": The guest selected a " NAME " sample rate"
" of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
" are supported.\n",
audio: log unimplemented audio device sample rates Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-01-21 10:47:25 +01:00
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 and truncation.
*/
samples = samples < SIZE_MAX ? samples + 1 : SIZE_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, hw->info.freq);
#else
sw->rate = st_rate_start(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->total_hw_samples_mixed = 0;
sw->empty = 1;
#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_PIPEWIRE
case AUDIODEV_DRIVER_PIPEWIRE:
return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.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