uaudio and Digigram UAX220
Kazuhito HONDA
kazuhito at ph.noda.tus.ac.jp
Mon Nov 7 12:04:37 PST 2005
Sorry, I forgot to attach a patch for uaudio.h.
I'll attach a new patch including
changes of uaudio.h and fixes of some typos.
From: Julian Elischer <julian at elischer.org>
Subject: Re: uaudio and Digigram UAX220
Date: Mon, 07 Nov 2005 09:29:34 -0800
> Does it apply to 6?
`patched_6_patch' was made on FreeBSD 6.0 which a sound patch was applied.
(http://people.freebsd.org/~ariff/snd_RELENG_6_0_20051030_058.diff)
I checked that this patch was available on patched FreeBSD 6-stable
(http://people.freebsd.org/~ariff/snd_RELENG_6_20051030_058.diff).
`normal_6_patch' was made from FreeBSD 6 without the sound patch.
I don't test it, but I'm sure that it is available.
sincerely yours,
Kazuhito HONDA
-------------- next part --------------
--- uaudio.c.old Wed Nov 2 15:31:54 2005
+++ uaudio.c Tue Nov 8 03:18:06 2005
@@ -239,6 +239,7 @@ struct uaudio_softc {
#define HAS_MULAW 0x10
#define UA_NOFRAC 0x20 /* don't do sample rate adjustment */
#define HAS_24 0x40
+#define HAS_32 0x80
int sc_mode; /* play/record capability */
struct mixerctl *sc_ctls; /* mixer controls */
int sc_nctls; /* # of mixer controls */
@@ -2050,7 +2051,7 @@ uaudio_process_as(struct uaudio_softc *s
format = UGETW(asid->wFormatTag);
chan = asf1d->bNrChannels;
prec = asf1d->bBitResolution;
- if (prec != 8 && prec != 16 && prec != 24) {
+ if (prec != 8 && prec != 16 && prec != 24 && prec != 32) {
printf("%s: ignored setting with precision %d\n",
USBDEVNAME(sc->sc_dev), prec);
return USBD_NORMAL_COMPLETION;
@@ -2063,6 +2064,8 @@ uaudio_process_as(struct uaudio_softc *s
sc->sc_altflags |= HAS_16;
} else if (prec == 24) {
sc->sc_altflags |= HAS_24;
+ } else if (prec == 32) {
+ sc->sc_altflags |= HAS_32;
}
enc = AUDIO_ENCODING_SLINEAR_LE;
format_str = "pcm";
@@ -3742,7 +3745,7 @@ uaudio_init_params(struct uaudio_softc *
if ((sc->sc_playchan.pipe != NULL) || (sc->sc_recchan.pipe != NULL))
return (-1);
- switch(ch->format & 0x0000FFFF) {
+ switch(ch->format & 0x000FFFFF) {
case AFMT_U8:
enc = AUDIO_ENCODING_ULINEAR_LE;
ch->precision = 8;
@@ -3775,6 +3778,38 @@ uaudio_init_params(struct uaudio_softc *
enc = AUDIO_ENCODING_ULINEAR_BE;
ch->precision = 16;
break;
+ case AFMT_S24_LE:
+ enc = AUDIO_ENCODING_SLINEAR_LE;
+ ch->precision = 24;
+ break;
+ case AFMT_S24_BE:
+ enc = AUDIO_ENCODING_SLINEAR_BE;
+ ch->precision = 24;
+ break;
+ case AFMT_U24_LE:
+ enc = AUDIO_ENCODING_ULINEAR_LE;
+ ch->precision = 24;
+ break;
+ case AFMT_U24_BE:
+ enc = AUDIO_ENCODING_ULINEAR_BE;
+ ch->precision = 24;
+ break;
+ case AFMT_S32_LE:
+ enc = AUDIO_ENCODING_SLINEAR_LE;
+ ch->precision = 32;
+ break;
+ case AFMT_S32_BE:
+ enc = AUDIO_ENCODING_SLINEAR_BE;
+ ch->precision = 32;
+ break;
+ case AFMT_U32_LE:
+ enc = AUDIO_ENCODING_ULINEAR_LE;
+ ch->precision = 32;
+ break;
+ case AFMT_U32_BE:
+ enc = AUDIO_ENCODING_ULINEAR_BE;
+ ch->precision = 32;
+ break;
default:
enc = 0;
ch->precision = 16;
@@ -3857,83 +3892,135 @@ uaudio_init_params(struct uaudio_softc *
return (0);
}
-void
-uaudio_query_formats(device_t dev, u_int32_t *pfmt, u_int32_t *rfmt)
+struct uaudio_conversion {
+ uint8_t uaudio_fmt;
+ uint8_t uaudio_prec;
+ uint32_t freebsd_fmt;
+};
+
+const struct uaudio_conversion const accepted_conversion[] = {
+ {AUDIO_ENCODING_ULINEAR_LE, 8, AFMT_U8},
+ {AUDIO_ENCODING_ULINEAR_LE, 16, AFMT_U16_LE},
+ {AUDIO_ENCODING_ULINEAR_LE, 24, AFMT_U24_LE},
+ {AUDIO_ENCODING_ULINEAR_LE, 32, AFMT_U32_LE},
+ {AUDIO_ENCODING_ULINEAR_BE, 16, AFMT_U16_BE},
+ {AUDIO_ENCODING_ULINEAR_BE, 24, AFMT_U24_BE},
+ {AUDIO_ENCODING_ULINEAR_BE, 32, AFMT_U32_BE},
+ {AUDIO_ENCODING_SLINEAR_LE, 8, AFMT_S8},
+ {AUDIO_ENCODING_SLINEAR_LE, 16, AFMT_S16_LE},
+ {AUDIO_ENCODING_SLINEAR_LE, 24, AFMT_S24_LE},
+ {AUDIO_ENCODING_SLINEAR_LE, 24, AFMT_S32_LE},
+ {AUDIO_ENCODING_SLINEAR_BE, 16, AFMT_S16_BE},
+ {AUDIO_ENCODING_SLINEAR_BE, 24, AFMT_S24_BE},
+ {AUDIO_ENCODING_SLINEAR_BE, 24, AFMT_S32_BE},
+ {AUDIO_ENCODING_ALAW, 8, AFMT_A_LAW},
+ {AUDIO_ENCODING_ULAW, 8, AFMT_MU_LAW},
+ {0,0,0}
+};
+
+unsigned
+uaudio_query_formats(device_t dev, int reqdir, unsigned maxfmt, struct pcmchan_caps *cap)
{
- int i, pn=0, rn=0;
- int prec, dir;
- u_int32_t fmt;
struct uaudio_softc *sc;
-
- const struct usb_audio_streaming_type1_descriptor *a1d;
+ const struct usb_audio_streaming_type1_descriptor *asf1d;
+ const struct uaudio_conversion *iterator;
+ unsigned fmtcount, foundcount;
+ u_int32_t fmt;
+ uint8_t format, numchan, subframesize, prec, dir, iscontinuous;
+ int freq, freq_min, freq_max;
+ char *numchannel_descr;
+ char freq_descr[64];
+ int i,r;
sc = device_get_softc(dev);
+ if (sc == NULL)
+ return 0;
+
+ cap->minspeed = cap->maxspeed = 0;
+ foundcount = fmtcount = 0;
for (i = 0; i < sc->sc_nalts; i++) {
- fmt = 0;
- a1d = sc->sc_alts[i].asf1desc;
- prec = a1d->bBitResolution; /* precision */
+ dir = UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
- switch (sc->sc_alts[i].encoding) {
- case AUDIO_ENCODING_ULINEAR_LE:
- if (prec == 8) {
- fmt = AFMT_U8;
- } else if (prec == 16) {
- fmt = AFMT_U16_LE;
- }
- break;
- case AUDIO_ENCODING_SLINEAR_LE:
- if (prec == 8) {
- fmt = AFMT_S8;
- } else if (prec == 16) {
- fmt = AFMT_S16_LE;
- }
- break;
- case AUDIO_ENCODING_ULINEAR_BE:
- if (prec == 16) {
- fmt = AFMT_U16_BE;
- }
- break;
- case AUDIO_ENCODING_SLINEAR_BE:
- if (prec == 16) {
- fmt = AFMT_S16_BE;
- }
- break;
- case AUDIO_ENCODING_ALAW:
- if (prec == 8) {
- fmt = AFMT_A_LAW;
- }
- break;
- case AUDIO_ENCODING_ULAW:
- if (prec == 8) {
- fmt = AFMT_MU_LAW;
- }
- break;
- }
+ if ((dir == UE_DIR_OUT) != (reqdir == PCMDIR_PLAY))
+ continue;
- if (fmt != 0) {
- if (a1d->bNrChannels == 2) { /* stereo/mono */
- fmt |= AFMT_STEREO;
- } else if (a1d->bNrChannels != 1) {
- fmt = 0;
- }
+ asf1d = sc->sc_alts[i].asf1desc;
+ format = sc->sc_alts[i].encoding;
+
+ numchan = asf1d->bNrChannels;
+ subframesize = asf1d->bSubFrameSize;
+ prec = asf1d->bBitResolution; /* precision */
+ iscontinuous = asf1d->bSamFreqType == UA_SAMP_CONTNUOUS;
+
+ if (iscontinuous)
+ snprintf(freq_descr, sizeof(freq_descr), "continous min %d max %d", UA_SAMP_LO(asf1d), UA_SAMP_HI(asf1d));
+ else
+ snprintf(freq_descr, sizeof(freq_descr), "fixed frequency (%d listed formats)", asf1d->bSamFreqType);
+
+ if (numchan == 1)
+ numchannel_descr = " (mono)";
+ else if (numchan == 2)
+ numchannel_descr = " (stereo)";
+ else
+ numchannel_descr = "";
+
+ if (bootverbose) {
+ device_printf(dev, "uaudio_query_formats: found a native %s channel%s %s %dbit %dbytes/subframe X %d channels = %d bytes per sample\n",
+ (dir==UE_DIR_OUT)?"playback":"record",
+ numchannel_descr, freq_descr,
+ prec, subframesize, numchan, subframesize*numchan);
}
+ /*
+ * Now start rejecting the ones that don't map to FreeBSD
+ */
- if (fmt != 0) {
- dir= UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
- if (dir == UE_DIR_OUT) {
- pfmt[pn++] = fmt;
- } else if (dir == UE_DIR_IN) {
- rfmt[rn++] = fmt;
+ if (numchan != 1 && numchan != 2)
+ continue;
+
+ for (iterator = accepted_conversion ; iterator->uaudio_fmt != 0 ; iterator++)
+ if (iterator->uaudio_fmt == format && iterator->uaudio_prec == prec)
+ break;
+
+ if (iterator->uaudio_fmt == 0)
+ continue;
+
+ fmt = iterator->freebsd_fmt;
+
+ if (numchan == 2)
+ fmt |= AFMT_STEREO;
+
+ foundcount++;
+
+ if (fmtcount >= maxfmt)
+ continue;
+
+ cap->fmtlist[fmtcount++] = fmt;
+
+ if (iscontinuous) {
+ freq_min = UA_SAMP_LO(asf1d);
+ freq_max = UA_SAMP_HI(asf1d);
+
+ if (cap->minspeed == 0 || freq_min < cap->minspeed)
+ cap->minspeed = freq_min;
+ if (cap->maxspeed == 0)
+ cap->maxspeed = cap->minspeed;
+ if (freq_max > cap->maxspeed)
+ cap->maxspeed = freq_max;
+ } else {
+ for (r = 0; r < asf1d->bSamFreqType; r++) {
+ freq = UA_GETSAMP(asf1d, r);
+ if (cap->minspeed == 0 || freq < cap->minspeed)
+ cap->minspeed = freq;
+ if (cap->maxspeed == 0)
+ cap->maxspeed = cap->minspeed;
+ if (freq > cap->maxspeed)
+ cap->maxspeed = freq;
}
}
-
- if ((pn > 8*2) || (rn > 8*2))
- break;
}
- pfmt[pn] = 0;
- rfmt[rn] = 0;
- return;
+ cap->fmtlist[fmtcount] = 0;
+ return foundcount;
}
void
@@ -3982,25 +4069,81 @@ uaudio_chan_set_param_blocksize(device_t
return;
}
-void
-uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int dir)
+int
+uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int reqdir)
{
+ const struct uaudio_conversion *iterator;
struct uaudio_softc *sc;
struct chan *ch;
+ int i, r, score, hiscore, bestspeed;
sc = device_get_softc(dev);
#ifndef NO_RECORDING
- if (dir == PCMDIR_PLAY)
+ if (reqdir == PCMDIR_PLAY)
ch = &sc->sc_playchan;
else
ch = &sc->sc_recchan;
#else
ch = &sc->sc_playchan;
#endif
+ /*
+ * We are successful if we find an endpoint that matches our selected format and it
+ * supports the requested speed.
+ */
+ hiscore = 0;
+ bestspeed = 1;
+ for (i = 0; i < sc->sc_nalts; i++) {
+ int dir = UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
+ int format = sc->sc_alts[i].encoding;
+ const struct usb_audio_streaming_type1_descriptor *asf1d = sc->sc_alts[i].asf1desc;
+ int iscontinuous = asf1d->bSamFreqType == UA_SAMP_CONTNUOUS;
- ch->sample_rate = speed;
+ if ((dir == UE_DIR_OUT) != (reqdir == PCMDIR_PLAY))
+ continue;
- return;
+ for (iterator = accepted_conversion ; iterator->uaudio_fmt != 0 ; iterator++)
+ if (iterator->uaudio_fmt != format || iterator->freebsd_fmt != (ch->format&0xfffffff))
+ continue;
+ if (iscontinuous) {
+ if (speed >= UA_SAMP_LO(asf1d) && speed <= UA_SAMP_HI(asf1d)) {
+ ch->sample_rate = speed;
+ return speed;
+ } else if (speed < UA_SAMP_LO(asf1d)) {
+ score = 0xfff * speed / UA_SAMP_LO(asf1d);
+ if (score > hiscore) {
+ bestspeed = UA_SAMP_LO(asf1d);
+ hiscore = score;
+ }
+ } else if (speed < UA_SAMP_HI(asf1d)) {
+ score = 0xfff * UA_SAMP_HI(asf1d) / speed;
+ if (score > hiscore) {
+ bestspeed = UA_SAMP_HI(asf1d);
+ hiscore = score;
+ }
+ }
+ continue;
+ }
+ for (r = 0; r < asf1d->bSamFreqType; r++) {
+ if (speed == UA_GETSAMP(asf1d, r)) {
+ ch->sample_rate = speed;
+ return speed;
+ }
+ if (speed > UA_GETSAMP(asf1d, r))
+ score = 0xfff * UA_GETSAMP(asf1d, r) / speed;
+ else
+ score = 0xfff * speed / UA_GETSAMP(asf1d, r);
+ if (score > hiscore) {
+ bestspeed = UA_GETSAMP(asf1d, r);
+ hiscore = score;
+ }
+ }
+ }
+ if (bestspeed != 1) {
+ ch->sample_rate = bestspeed;
+ return bestspeed;
+ }
+
+ return 0;
}
int
--- uaudio.h.old Thu Apr 28 02:16:27 2005
+++ uaudio.h Tue Nov 8 03:18:51 2005
@@ -41,7 +41,7 @@ int uaudio_halt_in_dma(device_t dev);
#endif
void uaudio_chan_set_param(device_t, u_char *, u_char *);
void uaudio_chan_set_param_blocksize(device_t dev, u_int32_t blocksize, int dir);
-void uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int dir);
+int uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int reqdir);
void uaudio_chan_set_param_format(device_t dev, u_int32_t format,int dir);
int uaudio_chan_getptr(device_t dev, int);
void uaudio_mixer_set(device_t dev, unsigned type, unsigned left,
@@ -49,5 +49,5 @@ void uaudio_mixer_set(device_t dev, unsi
u_int32_t uaudio_mixer_setrecsrc(device_t dev, u_int32_t src);
u_int32_t uaudio_query_mix_info(device_t dev);
u_int32_t uaudio_query_recsrc_info(device_t dev);
-void uaudio_query_formats(device_t dev, u_int32_t *pfmt, u_int32_t *rfmt);
+unsigned uaudio_query_formats(device_t dev, int dir, unsigned maxfmt, struct pcmchan_caps *fmt);
void uaudio_sndstat_register(device_t dev);
--- uaudio_pcm.c.old Wed Nov 2 15:31:54 2005
+++ uaudio_pcm.c Tue Nov 8 03:18:17 2005
@@ -49,16 +49,13 @@ struct ua_info {
device_t sc_dev;
u_int32_t bufsz;
struct ua_chinfo pch, rch;
+#define FORMAT_NUM 32
+ u_int32_t ua_playfmt[FORMAT_NUM*2+1]; /* FORMAT_NUM format * (stereo or mono) + endptr */
+ u_int32_t ua_recfmt[FORMAT_NUM*2+1]; /* FORMAT_NUM format * (stereo or mono) + endptr */
+ struct pcmchan_caps ua_playcaps;
+ struct pcmchan_caps ua_reccaps;
};
-static u_int32_t ua_playfmt[8*2+1]; /* 8 format * (stereo or mono) + endptr */
-
-static struct pcmchan_caps ua_playcaps = {8000, 48000, ua_playfmt, 0};
-
-static u_int32_t ua_recfmt[8*2+1]; /* 8 format * (stereo or mono) + endptr */
-
-static struct pcmchan_caps ua_reccaps = {8000, 48000, ua_recfmt, 0};
-
#define UAUDIO_DEFAULT_BUFSZ 16*1024
/************************************************************/
@@ -76,23 +73,9 @@ ua_chan_init(kobj_t obj, void *devinfo,
ch->dir = dir;
pa_dev = device_get_parent(sc->sc_dev);
- /* Create ua_playfmt[] & ua_recfmt[] */
- uaudio_query_formats(pa_dev, (u_int32_t *)&ua_playfmt, (u_int32_t *)&ua_recfmt);
- if (dir == PCMDIR_PLAY) {
- if (ua_playfmt[0] == 0) {
- printf("play channel supported format list invalid\n");
- return NULL;
- }
- } else {
- if (ua_recfmt[0] == 0) {
- printf("record channel supported format list invalid\n");
- return NULL;
- }
-
- }
ch->buf = malloc(sc->bufsz, M_DEVBUF, M_NOWAIT);
- if (ch->buf == NULL)
+ if (ch->buf == NULL)
return NULL;
if (sndbuf_setup(b, ch->buf, sc->bufsz) != 0) {
free(ch->buf, M_DEVBUF);
@@ -133,6 +116,9 @@ ua_chan_setformat(kobj_t obj, void *data
struct ua_chinfo *ch = data;
+ /*
+ * At this point, no need to query as we shouldn't select an unsorted format
+ */
ua = ch->parent;
pa_dev = device_get_parent(ua->sc_dev);
uaudio_chan_set_param_format(pa_dev, format, ch->dir);
@@ -144,15 +130,15 @@ ua_chan_setformat(kobj_t obj, void *data
static int
ua_chan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
+ struct ua_chinfo *ch;
device_t pa_dev;
- struct ua_info *ua;
+ int bestspeed;
- struct ua_chinfo *ch = data;
- ch->spd = speed;
+ ch = data;
+ pa_dev = device_get_parent(ch->parent->sc_dev);
- ua = ch->parent;
- pa_dev = device_get_parent(ua->sc_dev);
- uaudio_chan_set_param_speed(pa_dev, speed, ch->dir);
+ if ((bestspeed = uaudio_chan_set_param_speed(pa_dev, speed, ch->dir)))
+ ch->spd = bestspeed;
return ch->spd;
}
@@ -224,9 +210,10 @@ ua_chan_getptr(kobj_t obj, void *data)
static struct pcmchan_caps *
ua_chan_getcaps(kobj_t obj, void *data)
{
- struct ua_chinfo *ch = data;
+ struct ua_chinfo *ch;
- return (ch->dir == PCMDIR_PLAY) ? &ua_playcaps : & ua_reccaps;
+ ch = data;
+ return (ch->dir == PCMDIR_PLAY) ? &(ch->parent->ua_playcaps) : &(ch->parent->ua_reccaps);
}
static kobj_method_t ua_chan_methods[] = {
@@ -327,42 +314,63 @@ ua_attach(device_t dev)
{
struct ua_info *ua;
char status[SND_STATUSLEN];
+ device_t pa_dev;
+ u_int32_t nplay, nrec;
+ int i;
- ua = (struct ua_info *)malloc(sizeof *ua, M_DEVBUF, M_NOWAIT);
- if (!ua)
+ ua = (struct ua_info *)malloc(sizeof *ua, M_DEVBUF, M_ZERO | M_NOWAIT);
+ if (ua == NULL)
return ENXIO;
- bzero(ua, sizeof *ua);
ua->sc_dev = dev;
+ pa_dev = device_get_parent(dev);
+
ua->bufsz = pcm_getbuffersize(dev, 4096, UAUDIO_DEFAULT_BUFSZ, 65536);
if (bootverbose)
device_printf(dev, "using a default buffer size of %jd\n", (intmax_t)ua->bufsz);
if (mixer_init(dev, &ua_mixer_class, ua)) {
- return(ENXIO);
+ goto bad;
}
snprintf(status, SND_STATUSLEN, "at ? %s", PCM_KLDSTRING(snd_uaudio));
+ ua->ua_playcaps.fmtlist = ua->ua_playfmt;
+ ua->ua_reccaps.fmtlist = ua->ua_recfmt;
+ nplay = uaudio_query_formats(pa_dev, PCMDIR_PLAY, FORMAT_NUM * 2, &ua->ua_playcaps);
+ nrec = uaudio_query_formats(pa_dev, PCMDIR_REC, FORMAT_NUM * 2, &ua->ua_reccaps);
+
+ if (nplay > 1)
+ nplay = 1;
+ if (nrec > 1)
+ nrec = 1;
+
#ifndef NO_RECORDING
- if (pcm_register(dev, ua, 1, 1)) {
+ if (pcm_register(dev, ua, nplay, nrec)) {
#else
- if (pcm_register(dev, ua, 1, 0)) {
+ if (pcm_register(dev, ua, nplay, 0)) {
#endif
- return(ENXIO);
+ goto bad;
}
sndstat_unregister(dev);
uaudio_sndstat_register(dev);
- pcm_addchan(dev, PCMDIR_PLAY, &ua_chan_class, ua);
+ for (i = 0; i < nplay; i++) {
+ pcm_addchan(dev, PCMDIR_PLAY, &ua_chan_class, ua);
+ }
#ifndef NO_RECORDING
- pcm_addchan(dev, PCMDIR_REC, &ua_chan_class, ua);
+ for (i = 0; i < nrec; i++) {
+ pcm_addchan(dev, PCMDIR_REC, &ua_chan_class, ua);
+ }
#endif
pcm_setstatus(dev, status);
return 0;
+
+bad: free(ua, M_DEVBUF);
+ return ENXIO;
}
static int
-------------- next part --------------
--- uaudio.c.orig Tue Nov 8 04:17:47 2005
+++ uaudio.c Tue Nov 8 04:50:46 2005
@@ -231,6 +231,7 @@ struct uaudio_softc {
#define HAS_MULAW 0x10
#define UA_NOFRAC 0x20 /* don't do sample rate adjustment */
#define HAS_24 0x40
+#define HAS_32 0x80
int sc_mode; /* play/record capability */
struct mixerctl *sc_ctls; /* mixer controls */
int sc_nctls; /* # of mixer controls */
@@ -2027,7 +2028,7 @@ uaudio_process_as(struct uaudio_softc *s
format = UGETW(asid->wFormatTag);
chan = asf1d->bNrChannels;
prec = asf1d->bBitResolution;
- if (prec != 8 && prec != 16 && prec != 24) {
+ if (prec != 8 && prec != 16 && prec != 24 && prec != 32) {
printf("%s: ignored setting with precision %d\n",
USBDEVNAME(sc->sc_dev), prec);
return (USBD_NORMAL_COMPLETION);
@@ -2040,6 +2041,8 @@ uaudio_process_as(struct uaudio_softc *s
sc->sc_altflags |= HAS_16;
} else if (prec == 24) {
sc->sc_altflags |= HAS_24;
+ } else if (prec == 32) {
+ sc->sc_altflags |= HAS_32;
}
enc = AUDIO_ENCODING_SLINEAR_LE;
format_str = "pcm";
@@ -3690,7 +3693,7 @@ uaudio_init_params(struct uaudio_softc *
if ((sc->sc_playchan.pipe != NULL) || (sc->sc_recchan.pipe != NULL))
return (-1);
- switch(ch->format & 0x0000FFFF) {
+ switch(ch->format & 0x000FFFFF) {
case AFMT_U8:
enc = AUDIO_ENCODING_ULINEAR_LE;
ch->precision = 8;
@@ -3723,6 +3726,38 @@ uaudio_init_params(struct uaudio_softc *
enc = AUDIO_ENCODING_ULINEAR_BE;
ch->precision = 16;
break;
+ case AFMT_S24_LE:
+ enc = AUDIO_ENCODING_SLINEAR_LE;
+ ch->precision = 24;
+ break;
+ case AFMT_S24_BE:
+ enc = AUDIO_ENCODING_SLINEAR_BE;
+ ch->precision = 24;
+ break;
+ case AFMT_U24_LE:
+ enc = AUDIO_ENCODING_ULINEAR_LE;
+ ch->precision = 24;
+ break;
+ case AFMT_U24_BE:
+ enc = AUDIO_ENCODING_ULINEAR_BE;
+ ch->precision = 24;
+ break;
+ case AFMT_S32_LE:
+ enc = AUDIO_ENCODING_SLINEAR_LE;
+ ch->precision = 32;
+ break;
+ case AFMT_S32_BE:
+ enc = AUDIO_ENCODING_SLINEAR_BE;
+ ch->precision = 32;
+ break;
+ case AFMT_U32_LE:
+ enc = AUDIO_ENCODING_ULINEAR_LE;
+ ch->precision = 32;
+ break;
+ case AFMT_U32_BE:
+ enc = AUDIO_ENCODING_ULINEAR_BE;
+ ch->precision = 32;
+ break;
default:
enc = 0;
ch->precision = 16;
@@ -3805,83 +3840,135 @@ uaudio_init_params(struct uaudio_softc *
return (0);
}
-void
-uaudio_query_formats(device_t dev, u_int32_t *pfmt, u_int32_t *rfmt)
+struct uaudio_conversion {
+ uint8_t uaudio_fmt;
+ uint8_t uaudio_prec;
+ uint32_t freebsd_fmt;
+};
+
+const struct uaudio_conversion const accepted_conversion[] = {
+ {AUDIO_ENCODING_ULINEAR_LE, 8, AFMT_U8},
+ {AUDIO_ENCODING_ULINEAR_LE, 16, AFMT_U16_LE},
+ {AUDIO_ENCODING_ULINEAR_LE, 24, AFMT_U24_LE},
+ {AUDIO_ENCODING_ULINEAR_LE, 32, AFMT_U32_LE},
+ {AUDIO_ENCODING_ULINEAR_BE, 16, AFMT_U16_BE},
+ {AUDIO_ENCODING_ULINEAR_BE, 24, AFMT_U24_BE},
+ {AUDIO_ENCODING_ULINEAR_BE, 32, AFMT_U32_BE},
+ {AUDIO_ENCODING_SLINEAR_LE, 8, AFMT_S8},
+ {AUDIO_ENCODING_SLINEAR_LE, 16, AFMT_S16_LE},
+ {AUDIO_ENCODING_SLINEAR_LE, 24, AFMT_S24_LE},
+ {AUDIO_ENCODING_SLINEAR_LE, 24, AFMT_S32_LE},
+ {AUDIO_ENCODING_SLINEAR_BE, 16, AFMT_S16_BE},
+ {AUDIO_ENCODING_SLINEAR_BE, 24, AFMT_S24_BE},
+ {AUDIO_ENCODING_SLINEAR_BE, 24, AFMT_S32_BE},
+ {AUDIO_ENCODING_ALAW, 8, AFMT_A_LAW},
+ {AUDIO_ENCODING_ULAW, 8, AFMT_MU_LAW},
+ {0,0,0}
+};
+
+unsigned
+uaudio_query_formats(device_t dev, int reqdir, unsigned maxfmt, struct pcmchan_caps *cap)
{
- int i, pn=0, rn=0;
- int prec, dir;
- u_int32_t fmt;
struct uaudio_softc *sc;
-
- const struct usb_audio_streaming_type1_descriptor *a1d;
+ const struct usb_audio_streaming_type1_descriptor *asf1d;
+ const struct uaudio_conversion *iterator;
+ unsigned fmtcount, foundcount;
+ u_int32_t fmt;
+ uint8_t format, numchan, subframesize, prec, dir, iscontinuous;
+ int freq, freq_min, freq_max;
+ char *numchannel_descr;
+ char freq_descr[64];
+ int i,r;
sc = device_get_softc(dev);
+ if (sc == NULL)
+ return 0;
+
+ cap->minspeed = cap->maxspeed = 0;
+ foundcount = fmtcount = 0;
for (i = 0; i < sc->sc_nalts; i++) {
- fmt = 0;
- a1d = sc->sc_alts[i].asf1desc;
- prec = a1d->bBitResolution; /* precision */
+ dir = UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
- switch (sc->sc_alts[i].encoding) {
- case AUDIO_ENCODING_ULINEAR_LE:
- if (prec == 8) {
- fmt = AFMT_U8;
- } else if (prec == 16) {
- fmt = AFMT_U16_LE;
- }
- break;
- case AUDIO_ENCODING_SLINEAR_LE:
- if (prec == 8) {
- fmt = AFMT_S8;
- } else if (prec == 16) {
- fmt = AFMT_S16_LE;
- }
- break;
- case AUDIO_ENCODING_ULINEAR_BE:
- if (prec == 16) {
- fmt = AFMT_U16_BE;
- }
- break;
- case AUDIO_ENCODING_SLINEAR_BE:
- if (prec == 16) {
- fmt = AFMT_S16_BE;
- }
- break;
- case AUDIO_ENCODING_ALAW:
- if (prec == 8) {
- fmt = AFMT_A_LAW;
- }
- break;
- case AUDIO_ENCODING_ULAW:
- if (prec == 8) {
- fmt = AFMT_MU_LAW;
- }
- break;
- }
+ if ((dir == UE_DIR_OUT) != (reqdir == PCMDIR_PLAY))
+ continue;
- if (fmt != 0) {
- if (a1d->bNrChannels == 2) { /* stereo/mono */
- fmt |= AFMT_STEREO;
- } else if (a1d->bNrChannels != 1) {
- fmt = 0;
- }
+ asf1d = sc->sc_alts[i].asf1desc;
+ format = sc->sc_alts[i].encoding;
+
+ numchan = asf1d->bNrChannels;
+ subframesize = asf1d->bSubFrameSize;
+ prec = asf1d->bBitResolution; /* precision */
+ iscontinuous = asf1d->bSamFreqType == UA_SAMP_CONTNUOUS;
+
+ if (iscontinuous)
+ snprintf(freq_descr, sizeof(freq_descr), "continous min %d max %d", UA_SAMP_LO(asf1d), UA_SAMP_HI(asf1d));
+ else
+ snprintf(freq_descr, sizeof(freq_descr), "fixed frequency (%d listed formats)", asf1d->bSamFreqType);
+
+ if (numchan == 1)
+ numchannel_descr = " (mono)";
+ else if (numchan == 2)
+ numchannel_descr = " (stereo)";
+ else
+ numchannel_descr = "";
+
+ if (bootverbose) {
+ device_printf(dev, "uaudio_query_formats: found a native %s channel%s %s %dbit %dbytes/subframe X %d channels = %d bytes per sample\n",
+ (dir==UE_DIR_OUT)?"playback":"record",
+ numchannel_descr, freq_descr,
+ prec, subframesize, numchan, subframesize*numchan);
}
+ /*
+ * Now start rejecting the ones that don't map to FreeBSD
+ */
- if (fmt != 0) {
- dir= UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
- if (dir == UE_DIR_OUT) {
- pfmt[pn++] = fmt;
- } else if (dir == UE_DIR_IN) {
- rfmt[rn++] = fmt;
+ if (numchan != 1 && numchan != 2)
+ continue;
+
+ for (iterator = accepted_conversion ; iterator->uaudio_fmt != 0 ; iterator++)
+ if (iterator->uaudio_fmt == format && iterator->uaudio_prec == prec)
+ break;
+
+ if (iterator->uaudio_fmt == 0)
+ continue;
+
+ fmt = iterator->freebsd_fmt;
+
+ if (numchan == 2)
+ fmt |= AFMT_STEREO;
+
+ foundcount++;
+
+ if (fmtcount >= maxfmt)
+ continue;
+
+ cap->fmtlist[fmtcount++] = fmt;
+
+ if (iscontinuous) {
+ freq_min = UA_SAMP_LO(asf1d);
+ freq_max = UA_SAMP_HI(asf1d);
+
+ if (cap->minspeed == 0 || freq_min < cap->minspeed)
+ cap->minspeed = freq_min;
+ if (cap->maxspeed == 0)
+ cap->maxspeed = cap->minspeed;
+ if (freq_max > cap->maxspeed)
+ cap->maxspeed = freq_max;
+ } else {
+ for (r = 0; r < asf1d->bSamFreqType; r++) {
+ freq = UA_GETSAMP(asf1d, r);
+ if (cap->minspeed == 0 || freq < cap->minspeed)
+ cap->minspeed = freq;
+ if (cap->maxspeed == 0)
+ cap->maxspeed = cap->minspeed;
+ if (freq > cap->maxspeed)
+ cap->maxspeed = freq;
}
}
-
- if ((pn > 8*2) || (rn > 8*2))
- break;
}
- pfmt[pn] = 0;
- rfmt[rn] = 0;
- return;
+ cap->fmtlist[fmtcount] = 0;
+ return foundcount;
}
void
@@ -3930,25 +4017,81 @@ uaudio_chan_set_param_blocksize(device_t
return;
}
-void
-uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int dir)
+int
+uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int reqdir)
{
+ const struct uaudio_conversion *iterator;
struct uaudio_softc *sc;
struct chan *ch;
+ int i, r, score, hiscore, bestspeed;
sc = device_get_softc(dev);
#ifndef NO_RECORDING
- if (dir == PCMDIR_PLAY)
+ if (reqdir == PCMDIR_PLAY)
ch = &sc->sc_playchan;
else
ch = &sc->sc_recchan;
#else
ch = &sc->sc_playchan;
#endif
+ /*
+ * We are successful if we find an endpoint that matches our selected format and it
+ * supports the requested speed.
+ */
+ hiscore = 0;
+ bestspeed = 1;
+ for (i = 0; i < sc->sc_nalts; i++) {
+ int dir = UE_GET_DIR(sc->sc_alts[i].edesc->bEndpointAddress);
+ int format = sc->sc_alts[i].encoding;
+ const struct usb_audio_streaming_type1_descriptor *asf1d = sc->sc_alts[i].asf1desc;
+ int iscontinuous = asf1d->bSamFreqType == UA_SAMP_CONTNUOUS;
- ch->sample_rate = speed;
+ if ((dir == UE_DIR_OUT) != (reqdir == PCMDIR_PLAY))
+ continue;
- return;
+ for (iterator = accepted_conversion ; iterator->uaudio_fmt != 0 ; iterator++)
+ if (iterator->uaudio_fmt != format || iterator->freebsd_fmt != (ch->format&0xfffffff))
+ continue;
+ if (iscontinuous) {
+ if (speed >= UA_SAMP_LO(asf1d) && speed <= UA_SAMP_HI(asf1d)) {
+ ch->sample_rate = speed;
+ return speed;
+ } else if (speed < UA_SAMP_LO(asf1d)) {
+ score = 0xfff * speed / UA_SAMP_LO(asf1d);
+ if (score > hiscore) {
+ bestspeed = UA_SAMP_LO(asf1d);
+ hiscore = score;
+ }
+ } else if (speed < UA_SAMP_HI(asf1d)) {
+ score = 0xfff * UA_SAMP_HI(asf1d) / speed;
+ if (score > hiscore) {
+ bestspeed = UA_SAMP_HI(asf1d);
+ hiscore = score;
+ }
+ }
+ continue;
+ }
+ for (r = 0; r < asf1d->bSamFreqType; r++) {
+ if (speed == UA_GETSAMP(asf1d, r)) {
+ ch->sample_rate = speed;
+ return speed;
+ }
+ if (speed > UA_GETSAMP(asf1d, r))
+ score = 0xfff * UA_GETSAMP(asf1d, r) / speed;
+ else
+ score = 0xfff * speed / UA_GETSAMP(asf1d, r);
+ if (score > hiscore) {
+ bestspeed = UA_GETSAMP(asf1d, r);
+ hiscore = score;
+ }
+ }
+ }
+ if (bestspeed != 1) {
+ ch->sample_rate = bestspeed;
+ return bestspeed;
+ }
+
+ return 0;
}
int
--- uaudio.h.orig Tue Nov 8 04:17:58 2005
+++ uaudio.h Tue Nov 8 04:50:46 2005
@@ -41,7 +41,7 @@ int uaudio_halt_in_dma(device_t dev);
#endif
void uaudio_chan_set_param(device_t, u_char *, u_char *);
void uaudio_chan_set_param_blocksize(device_t dev, u_int32_t blocksize, int dir);
-void uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int dir);
+int uaudio_chan_set_param_speed(device_t dev, u_int32_t speed, int reqdir);
void uaudio_chan_set_param_format(device_t dev, u_int32_t format,int dir);
int uaudio_chan_getptr(device_t dev, int);
void uaudio_mixer_set(device_t dev, unsigned type, unsigned left,
@@ -49,5 +49,5 @@ void uaudio_mixer_set(device_t dev, unsi
u_int32_t uaudio_mixer_setrecsrc(device_t dev, u_int32_t src);
u_int32_t uaudio_query_mix_info(device_t dev);
u_int32_t uaudio_query_recsrc_info(device_t dev);
-void uaudio_query_formats(device_t dev, u_int32_t *pfmt, u_int32_t *rfmt);
+unsigned uaudio_query_formats(device_t dev, int dir, unsigned maxfmt, struct pcmchan_caps *fmt);
void uaudio_sndstat_register(device_t dev);
--- uaudio_pcm.c.orig Tue Nov 8 04:17:47 2005
+++ uaudio_pcm.c Tue Nov 8 04:54:47 2005
@@ -49,16 +49,13 @@ struct ua_info {
device_t sc_dev;
u_int32_t bufsz;
struct ua_chinfo pch, rch;
+#define FORMAT_NUM 32
+ u_int32_t ua_playfmt[FORMAT_NUM*2+1]; /* FORMAT_NUM format * (stereo or mono) + endptr */
+ u_int32_t ua_recfmt[FORMAT_NUM*2+1]; /* FORMAT_NUM format * (stereo or mono) + endptr */
+ struct pcmchan_caps ua_playcaps;
+ struct pcmchan_caps ua_reccaps;
};
-static u_int32_t ua_playfmt[8*2+1]; /* 8 format * (stereo or mono) + endptr */
-
-static struct pcmchan_caps ua_playcaps = {8000, 48000, ua_playfmt, 0};
-
-static u_int32_t ua_recfmt[8*2+1]; /* 8 format * (stereo or mono) + endptr */
-
-static struct pcmchan_caps ua_reccaps = {8000, 48000, ua_recfmt, 0};
-
#define UAUDIO_DEFAULT_BUFSZ 16*1024
/************************************************************/
@@ -76,23 +73,9 @@ ua_chan_init(kobj_t obj, void *devinfo,
ch->dir = dir;
pa_dev = device_get_parent(sc->sc_dev);
- /* Create ua_playfmt[] & ua_recfmt[] */
- uaudio_query_formats(pa_dev, (u_int32_t *)&ua_playfmt, (u_int32_t *)&ua_recfmt);
- if (dir == PCMDIR_PLAY) {
- if (ua_playfmt[0] == 0) {
- printf("play channel supported format list invalid\n");
- return NULL;
- }
- } else {
- if (ua_recfmt[0] == 0) {
- printf("record channel supported format list invalid\n");
- return NULL;
- }
-
- }
ch->buf = malloc(sc->bufsz, M_DEVBUF, M_NOWAIT);
- if (ch->buf == NULL)
+ if (ch->buf == NULL)
return NULL;
if (sndbuf_setup(b, ch->buf, sc->bufsz) != 0) {
free(ch->buf, M_DEVBUF);
@@ -133,6 +116,9 @@ ua_chan_setformat(kobj_t obj, void *data
struct ua_chinfo *ch = data;
+ /*
+ * At this point, no need to query as we shouldn't select an unsorted format
+ */
ua = ch->parent;
pa_dev = device_get_parent(ua->sc_dev);
uaudio_chan_set_param_format(pa_dev, format, ch->dir);
@@ -144,15 +130,15 @@ ua_chan_setformat(kobj_t obj, void *data
static int
ua_chan_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
+ struct ua_chinfo *ch;
device_t pa_dev;
- struct ua_info *ua;
+ int bestspeed;
- struct ua_chinfo *ch = data;
- ch->spd = speed;
+ ch = data;
+ pa_dev = device_get_parent(ch->parent->sc_dev);
- ua = ch->parent;
- pa_dev = device_get_parent(ua->sc_dev);
- uaudio_chan_set_param_speed(pa_dev, speed, ch->dir);
+ if ((bestspeed = uaudio_chan_set_param_speed(pa_dev, speed, ch->dir)))
+ ch->spd = bestspeed;
return ch->spd;
}
@@ -224,9 +210,10 @@ ua_chan_getptr(kobj_t obj, void *data)
static struct pcmchan_caps *
ua_chan_getcaps(kobj_t obj, void *data)
{
- struct ua_chinfo *ch = data;
+ struct ua_chinfo *ch;
- return (ch->dir == PCMDIR_PLAY) ? &ua_playcaps : & ua_reccaps;
+ ch = data;
+ return (ch->dir == PCMDIR_PLAY) ? &(ch->parent->ua_playcaps) : &(ch->parent->ua_reccaps);
}
static kobj_method_t ua_chan_methods[] = {
@@ -318,42 +305,63 @@ ua_attach(device_t dev)
{
struct ua_info *ua;
char status[SND_STATUSLEN];
+ device_t pa_dev;
+ u_int32_t nplay, nrec;
+ int i;
- ua = (struct ua_info *)malloc(sizeof *ua, M_DEVBUF, M_NOWAIT);
- if (!ua)
+ ua = (struct ua_info *)malloc(sizeof *ua, M_DEVBUF, M_ZERO | M_NOWAIT);
+ if (ua == NULL)
return ENXIO;
- bzero(ua, sizeof *ua);
ua->sc_dev = dev;
+ pa_dev = device_get_parent(dev);
+
ua->bufsz = pcm_getbuffersize(dev, 4096, UAUDIO_DEFAULT_BUFSZ, 65536);
if (bootverbose)
device_printf(dev, "using a default buffer size of %jd\n", (intmax_t)ua->bufsz);
if (mixer_init(dev, &ua_mixer_class, ua)) {
- return(ENXIO);
+ goto bad;
}
snprintf(status, SND_STATUSLEN, "at addr ?");
+ ua->ua_playcaps.fmtlist = ua->ua_playfmt;
+ ua->ua_reccaps.fmtlist = ua->ua_recfmt;
+ nplay = uaudio_query_formats(pa_dev, PCMDIR_PLAY, FORMAT_NUM * 2, &ua->ua_playcaps);
+ nrec = uaudio_query_formats(pa_dev, PCMDIR_REC, FORMAT_NUM * 2, &ua->ua_reccaps);
+
+ if (nplay > 1)
+ nplay = 1;
+ if (nrec > 1)
+ nrec = 1;
+
#ifndef NO_RECORDING
- if (pcm_register(dev, ua, 1, 1)) {
+ if (pcm_register(dev, ua, nplay, nrec)) {
#else
- if (pcm_register(dev, ua, 1, 0)) {
+ if (pcm_register(dev, ua, nplay, 0)) {
#endif
- return(ENXIO);
+ goto bad;
}
sndstat_unregister(dev);
uaudio_sndstat_register(dev);
- pcm_addchan(dev, PCMDIR_PLAY, &ua_chan_class, ua);
+ for (i = 0; i < nplay; i++) {
+ pcm_addchan(dev, PCMDIR_PLAY, &ua_chan_class, ua);
+ }
#ifndef NO_RECORDING
- pcm_addchan(dev, PCMDIR_REC, &ua_chan_class, ua);
+ for (i = 0; i < nrec; i++) {
+ pcm_addchan(dev, PCMDIR_REC, &ua_chan_class, ua);
+ }
#endif
pcm_setstatus(dev, status);
return 0;
+
+bad: free(ua, M_DEVBUF);
+ return ENXIO;
}
static int
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