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[src/isaki-audio2]: src/sys/dev/pci Adapt to audio2.



details:   https://anonhg.NetBSD.org/src/rev/ae0a22567f41
branches:  isaki-audio2
changeset: 450889:ae0a22567f41
user:      isaki <isaki%NetBSD.org@localhost>
date:      Sun Apr 28 06:36:50 2019 +0000

description:
Adapt to audio2.
- Searching nearest frequency is unnecessary in audio2.

diffstat:

 sys/dev/pci/fms.c |  214 ++++++++++++-----------------------------------------
 1 files changed, 51 insertions(+), 163 deletions(-)

diffs (truncated from 302 to 300 lines):

diff -r e62b671d1e91 -r ae0a22567f41 sys/dev/pci/fms.c
--- a/sys/dev/pci/fms.c Sun Apr 28 05:07:00 2019 +0000
+++ b/sys/dev/pci/fms.c Sun Apr 28 06:36:50 2019 +0000
@@ -1,4 +1,4 @@
-/*     $NetBSD: fms.c,v 1.45.2.1 2019/04/21 05:11:22 isaki Exp $       */
+/*     $NetBSD: fms.c,v 1.45.2.2 2019/04/28 06:36:50 isaki Exp $       */
 
 /*-
  * Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
@@ -34,7 +34,7 @@
  */
 
 #include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: fms.c,v 1.45.2.1 2019/04/21 05:11:22 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: fms.c,v 1.45.2.2 2019/04/28 06:36:50 isaki Exp $");
 
 #include "mpu.h"
 
@@ -52,8 +52,6 @@
 #include <dev/pci/pcivar.h>
 
 #include <dev/audio_if.h>
-#include <dev/mulaw.h>
-#include <dev/auconv.h>
 
 #include <dev/ic/ac97var.h>
 #include <dev/ic/mpuvar.h>
@@ -75,10 +73,10 @@
 static void    fms_attach(device_t, device_t, void *);
 static int     fms_intr(void *);
 
-static int     fms_query_encoding(void *, struct audio_encoding *);
-static int     fms_set_params(void *, int, int, audio_params_t *,
-                              audio_params_t *, stream_filter_list_t *,
-                              stream_filter_list_t *);
+static int     fms_query_format(void *, audio_format_query_t *);
+static int     fms_set_format(void *, int,
+                              const audio_params_t *, const audio_params_t *,
+                              audio_filter_reg_t *, audio_filter_reg_t *);
 static int     fms_round_blocksize(void *, int, int, const audio_params_t *);
 static int     fms_halt_output(void *);
 static int     fms_halt_input(void *);
@@ -88,8 +86,6 @@
 static int     fms_query_devinfo(void *, mixer_devinfo_t *);
 static void    *fms_malloc(void *, int, size_t);
 static void    fms_free(void *, void *, size_t);
-static size_t  fms_round_buffersize(void *, int, size_t);
-static paddr_t fms_mappage(void *, void *, off_t, int);
 static int     fms_get_props(void *);
 static int     fms_trigger_output(void *, void *, void *, int,
                                   void (*)(void *), void *,
@@ -108,10 +104,29 @@
        "fms"
 };
 
+/*
+ * The frequency list in this format is also referred from fms_rate2index().
+ * So don't rearrange or delete entries.
+ */
+static const struct audio_format fms_formats[] = {
+       {
+               .mode           = AUMODE_PLAY | AUMODE_RECORD,
+               .encoding       = AUDIO_ENCODING_SLINEAR_LE,
+               .validbits      = 16,
+               .precision      = 16,
+               .channels       = 2,
+               .channel_mask   = AUFMT_STEREO,
+               .frequency_type = 11,
+               .frequency      = { 5500, 8000, 9600, 11025, 16000, 19200,
+                                   22050, 32000, 38400, 44100, 48000},
+       },
+};
+#define FMS_NFORMATS   __arraycount(fms_formats)
+
 
 static const struct audio_hw_if fms_hw_if = {
-       .query_encoding         = fms_query_encoding,
-       .set_params             = fms_set_params,
+       .query_format           = fms_query_format,
+       .set_format             = fms_set_format,
        .round_blocksize        = fms_round_blocksize,
        .halt_output            = fms_halt_output,
        .halt_input             = fms_halt_input,
@@ -121,8 +136,6 @@
        .query_devinfo          = fms_query_devinfo,
        .allocm                 = fms_malloc,
        .freem                  = fms_free,
-       .round_buffersize       = fms_round_buffersize,
-       .mappage                = fms_mappage,
        .get_props              = fms_get_props,
        .trigger_output         = fms_trigger_output,
        .trigger_input          = fms_trigger_input,
@@ -133,6 +146,7 @@
 static int     fms_read_codec(void *, uint8_t, uint16_t *);
 static int     fms_write_codec(void *, uint8_t, uint16_t);
 static int     fms_reset_codec(void *);
+static int     fms_rate2index(u_int);
 
 #define FM_PCM_VOLUME          0x00
 #define FM_FM_VOLUME           0x02
@@ -462,145 +476,45 @@
 }
 
 static int
-fms_query_encoding(void *addr, struct audio_encoding *fp)
+fms_query_format(void *addr, audio_format_query_t *afp)
 {
 
-       switch (fp->index) {
-       case 0:
-               strcpy(fp->name, AudioEmulaw);
-               fp->encoding = AUDIO_ENCODING_ULAW;
-               fp->precision = 8;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       case 1:
-               strcpy(fp->name, AudioEslinear_le);
-               fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
-               fp->precision = 16;
-               fp->flags = 0;
-               return 0;
-       case 2:
-               strcpy(fp->name, AudioEulinear);
-               fp->encoding = AUDIO_ENCODING_ULINEAR;
-               fp->precision = 8;
-               fp->flags = 0;
-               return 0;
-       case 3:
-               strcpy(fp->name, AudioEalaw);
-               fp->encoding = AUDIO_ENCODING_ALAW;
-               fp->precision = 8;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       case 4:
-               strcpy(fp->name, AudioEulinear_le);
-               fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
-               fp->precision = 16;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       case 5:
-               strcpy(fp->name, AudioEslinear);
-               fp->encoding = AUDIO_ENCODING_SLINEAR;
-               fp->precision = 8;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       case 6:
-               strcpy(fp->name, AudioEulinear_be);
-               fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
-               fp->precision = 16;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       case 7:
-               strcpy(fp->name, AudioEslinear_be);
-               fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
-               fp->precision = 16;
-               fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
-               return 0;
-       default:
-               return EINVAL;
-       }
+       return audio_query_format(fms_formats, FMS_NFORMATS, afp);
 }
 
-/*
- * Range below -limit- is set to -rate-
- * What a pity FM801 does not have 24000
- * 24000 -> 22050 sounds rather poor
- */
-static struct {
-       int limit;
-       int rate;
-} const fms_rates[11] = {
-       {  6600,  5500 },
-       {  8750,  8000 },
-       { 10250,  9600 },
-       { 13200, 11025 },
-       { 17500, 16000 },
-       { 20500, 19200 },
-       { 26500, 22050 },
-       { 35000, 32000 },
-       { 41000, 38400 },
-       { 46000, 44100 },
-       { 48000, 48000 },
-       /* anything above -> 48000 */
-};
+/* Return index number of sample_rate */
+static int
+fms_rate2index(u_int sample_rate)
+{
+       int i;
 
-#define FMS_NFORMATS   4
-#define FMS_FORMAT(enc, prec, ch, chmask) \
-       { \
-               .mode           = AUMODE_PLAY | AUMODE_RECORD, \
-               .encoding       = (enc), \
-               .validbits      = (prec), \
-               .precision      = (prec), \
-               .channels       = (ch), \
-               .channel_mask   = (chmask), \
-               .frequency_type = 11, \
-               .frequency      = { 5500, 8000, 9600, 11025, 16000, 19200, \
-                                   22050, 32000, 38400, 44100, 48000}, \
+       for (i = 0; i < fms_formats[0].frequency_type; i++) {
+               if (sample_rate == fms_formats[0].frequency[i])
+                       return i;
        }
-static const struct audio_format fms_formats[FMS_NFORMATS] = {
-       FMS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 2, AUFMT_STEREO),
-       FMS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 1, AUFMT_MONAURAL),
-       FMS_FORMAT(AUDIO_ENCODING_ULINEAR_LE,  8, 2, AUFMT_STEREO),
-       FMS_FORMAT(AUDIO_ENCODING_ULINEAR_LE,  8, 1, AUFMT_MONAURAL),
-};
+
+       /* NOTREACHED */
+       panic("fms_format.frequency mismatch?\n");
+}
 
 static int
-fms_set_params(void *addr, int setmode, int usemode,
-    audio_params_t *play, audio_params_t *rec, stream_filter_list_t *pfil,
-    stream_filter_list_t *rfil)
+fms_set_format(void *addr, int setmode,
+    const audio_params_t *play, const audio_params_t *rec,
+    audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
 {
        struct fms_softc *sc;
-       int i, index;
 
        sc = addr;
        if (setmode & AUMODE_PLAY) {
-               for (i = 0; i < 10 && play->sample_rate > fms_rates[i].limit;
-                    i++)
-                       continue;
-               play->sample_rate = fms_rates[i].rate;
-               index = auconv_set_converter(fms_formats, FMS_NFORMATS,
-                                            AUMODE_PLAY, play, FALSE, pfil);
-               if (index < 0)
-                       return EINVAL;
-               sc->sc_play_reg = i << 8;
-               if (fms_formats[index].channels == 2)
-                       sc->sc_play_reg |= FM_PLAY_STEREO;
-               if (fms_formats[index].precision == 16)
-                       sc->sc_play_reg |= FM_PLAY_16BIT;
+               sc->sc_play_reg = fms_rate2index(play->sample_rate) << 8;
+               sc->sc_play_reg |= FM_PLAY_STEREO;
+               sc->sc_play_reg |= FM_PLAY_16BIT;
        }
 
        if (setmode & AUMODE_RECORD) {
-               for (i = 0; i < 10 && rec->sample_rate > fms_rates[i].limit;
-                    i++)
-                       continue;
-               rec->sample_rate = fms_rates[i].rate;
-               index = auconv_set_converter(fms_formats, FMS_NFORMATS,
-                                            AUMODE_RECORD, rec, FALSE, rfil);
-               if (index < 0)
-                       return EINVAL;
-               sc->sc_rec_reg = i << 8;
-               if (fms_formats[index].channels == 2)
-                       sc->sc_rec_reg |= FM_REC_STEREO;
-               if (fms_formats[index].precision == 16)
-                       sc->sc_rec_reg |= FM_REC_16BIT;
+               sc->sc_rec_reg = fms_rate2index(rec->sample_rate) << 8;
+               sc->sc_rec_reg |= FM_REC_STEREO;
+               sc->sc_rec_reg |= FM_REC_16BIT;
        }
 
        return 0;
@@ -748,32 +662,6 @@
        panic("fms_free: trying to free unallocated memory");
 }
 
-static size_t
-fms_round_buffersize(void *addr, int direction, size_t size)
-{
-
-       return size;
-}
-
-static paddr_t
-fms_mappage(void *addr, void *mem, off_t off, int prot)
-{
-       struct fms_softc *sc;
-       struct fms_dma *p;
-
-       sc = addr;
-       if (off < 0)
-               return -1;
-
-       for (p = sc->sc_dmas; p && p->addr != mem; p = p->next)
-               continue;
-       if (p == NULL)
-               return -1;
-
-       return bus_dmamem_mmap(sc->sc_dmat, &p->seg, 1, off, prot,
-                              BUS_DMA_WAITOK);
-}
-
 static int



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