static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->in;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
g,
"qemu",
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
&error
);
if (!pa->stream) {
qpa_logerr (error, "pa_simple_new for capture failed\n");
goto fail1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto fail2;
}
if (audio_pt_init(&pa->pt, qpa_thread_in, hw, AUDIO_CAP, __func__)) {
goto fail3;
}
return 0;
fail3:
g_free (pa->pcm_buf);
pa->pcm_buf = NULL;
fail2:
if (pa->stream) {
pa_stream_unref (pa->stream);
pa->stream = NULL;
}
fail1:
return -1;
}
static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
int error;
static pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->s = FF(pa_simple_new) (
conf.server,
"qemu",
PA_STREAM_RECORD,
conf.source,
"pcm.capture",
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
&error
);
if (!pa->s) {
qpa_logerr (error, "pa_simple_new for capture failed\n");
goto fail1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto fail2;
}
if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
goto fail3;
}
return 0;
fail3:
qemu_free (pa->pcm_buf);
pa->pcm_buf = NULL;
fail2:
FF(pa_simple_free) (pa->s);
pa->s = NULL;
fail1:
return -1;
}
/*
* Prepare processing.
*/
void *st_rate_start (int inrate, int outrate)
{
struct rate *rate = audio_calloc (AUDIO_FUNC, 1, sizeof (*rate));
if (!rate) {
dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
return NULL;
}
rate->opos = 0;
/* increment */
rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
rate->ipos = 0;
rate->ilast.l = 0;
rate->ilast.r = 0;
return rate;
}
static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
int error;
pa_sample_spec ss;
pa_buffer_attr ba;
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->out;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
/*
* qemu audio tick runs at 100 Hz (by default), so processing
* data chunks worth 10 ms of sound should be a good fit.
*/
ba.tlength = pa_usec_to_bytes (10 * 1000, &ss);
ba.minreq = pa_usec_to_bytes (5 * 1000, &ss);
ba.maxlength = -1;
ba.prebuf = -1;
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
g,
"qemu",
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
&error
);
if (!pa->stream) {
qpa_logerr (error, "pa_simple_new for playback failed\n");
goto fail1;
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto fail2;
}
if (audio_pt_init(&pa->pt, qpa_thread_out, hw, AUDIO_CAP, __func__)) {
goto fail3;
}
return 0;
fail3:
g_free (pa->pcm_buf);
pa->pcm_buf = NULL;
fail2:
if (pa->stream) {
pa_stream_unref (pa->stream);
pa->stream = NULL;
}
fail1:
return -1;
}
static int qesd_init_in (HWVoiceIn *hw, struct audsettings *as)
{
ESDVoiceIn *esd = (ESDVoiceIn *) hw;
struct audsettings obt_as = *as;
int esdfmt = ESD_STREAM | ESD_RECORD;
int result = -1;
/* shut down verbose debug spew */
if (!D_ACTIVE)
stdio_disable();
esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO;
switch (as->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
esdfmt |= ESD_BITS8;
obt_as.fmt = AUD_FMT_U8;
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
esdfmt |= ESD_BITS16;
obt_as.fmt = AUD_FMT_S16;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
dolog ("Will use 16 instead of 32 bit samples\n");
esdfmt |= ESD_BITS16;
obt_as.fmt = AUD_FMT_S16;
break;
}
obt_as.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!esd->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
goto exit;
}
esd->fd = FF(esd_record_stream) (esdfmt, as->freq, conf.adc_host, NULL);
if (esd->fd < 0) {
if (conf.adc_host == NULL) {
esd->fd = FF(esd_record_stream) (esdfmt, as->freq, "localhost", NULL);
}
if (esd->fd < 0) {
qesd_logerr (errno, "esd_record_stream failed\n");
goto fail2;
}
}
{
int flags;
flags = fcntl(esd->fd, F_GETFL);
fcntl(esd->fd, F_SETFL, flags | O_NONBLOCK);
}
result = 0; /* success */
goto exit;
fail2:
qemu_free (esd->pcm_buf);
esd->pcm_buf = NULL;
exit:
if (!D_ACTIVE)
stdio_enable();
return result;
}
static int winwave_init_in (HWVoiceIn *hw, struct audsettings *as)
{
int i;
int err;
MMRESULT mr;
WAVEFORMATEX wfx;
WaveVoiceIn *wave;
wave = (WaveVoiceIn *) hw;
InitializeCriticalSection (&wave->crit_sect);
err = waveformat_from_audio_settings (&wfx, as);
if (err) {
goto err0;
}
mr = waveInOpen (&wave->hwi, WAVE_MAPPER, &wfx,
(DWORD_PTR) winwave_callback_in,
(DWORD_PTR) wave, CALLBACK_FUNCTION);
if (mr != MMSYSERR_NOERROR) {
winwave_logerr (mr, "waveInOpen");
goto err1;
}
wave->hdrs = audio_calloc (AUDIO_FUNC, conf.dac_headers,
sizeof (*wave->hdrs));
if (!wave->hdrs) {
goto err2;
}
audio_pcm_init_info (&hw->info, as);
hw->samples = conf.adc_samples * conf.adc_headers;
wave->avail = 0;
wave->pcm_buf = audio_calloc (AUDIO_FUNC, conf.adc_samples,
conf.adc_headers << hw->info.shift);
if (!wave->pcm_buf) {
goto err3;
}
for (i = 0; i < conf.adc_headers; ++i) {
WAVEHDR *h = &wave->hdrs[i];
h->dwUser = 0;
h->dwBufferLength = conf.adc_samples << hw->info.shift;
h->lpData = advance (wave->pcm_buf, i * h->dwBufferLength);
h->dwFlags = 0;
mr = waveInPrepareHeader (wave->hwi, h, sizeof (*h));
if (mr != MMSYSERR_NOERROR) {
winwave_logerr (mr, "waveInPrepareHeader(%d)", i);
goto err4;
}
}
wave->paused = 1;
winwave_add_buffers (wave, hw->samples);
return 0;
err4:
g_free (wave->pcm_buf);
err3:
g_free (wave->hdrs);
err2:
winwave_anal_close_in (wave);
err1:
err0:
return -1;
}
