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authorTakashi Sakamoto <o-takashi@sakamocchi.jp>2014-12-09 00:10:42 +0900
committerTakashi Iwai <tiwai@suse.de>2014-12-10 10:47:37 +0100
commit5cd1d3f47a6321612a51ab88ffe8ef65120fcbe0 (patch)
treef2f537e9ec9185754441c37f7896788a76795312 /sound/firewire/oxfw/oxfw-stream.c
parent5b59d8098d2a3fa8ea4ad07b96f62c00c3b3e8d3 (diff)
downloadlinux-5cd1d3f47a6321612a51ab88ffe8ef65120fcbe0.tar.bz2
ALSA: oxfw: Change the way to make PCM rules/constraints
In previous commit, this driver can get to know stream formations at each supported sampling rates. This commit uses it to make PCM rules/constraints and obsoletes hard-coded rules/constraints. For this purpose, this commit adds 'struct snd_oxfw_stream_formation' and snd_oxfw_stream_parse_format() to parse data channel formation of data block. According to datasheet of OXFW970/971, they support 32.0kHz to 196.0kHz. As long as developers investigate, some devices are confirmed to have several formats for the same sampling rate. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound/firewire/oxfw/oxfw-stream.c')
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c268
1 files changed, 268 insertions, 0 deletions
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index ebd156f3e29d..17e3802e6ac2 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -8,6 +8,35 @@
#include "oxfw.h"
+#define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512
+
+/*
+ * According to datasheet of Oxford Semiconductor:
+ * OXFW970: 32.0/44.1/48.0/96.0 Khz, 8 audio channels I/O
+ * OXFW971: 32.0/44.1/48.0/88.2/96.0/192.0 kHz, 16 audio channels I/O, MIDI I/O
+ */
+static const unsigned int oxfw_rate_table[] = {
+ [0] = 32000,
+ [1] = 44100,
+ [2] = 48000,
+ [3] = 88200,
+ [4] = 96000,
+ [5] = 192000,
+};
+
+/*
+ * See Table 5.7 – Sampling frequency for Multi-bit Audio
+ * in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA)
+ */
+static const unsigned int avc_stream_rate_table[] = {
+ [0] = 0x02,
+ [1] = 0x03,
+ [2] = 0x04,
+ [3] = 0x0a,
+ [4] = 0x05,
+ [5] = 0x07,
+};
+
int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw)
{
int err;
@@ -78,3 +107,242 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw)
else
amdtp_stream_update(&oxfw->rx_stream);
}
+
+/*
+ * See Table 6.16 - AM824 Stream Format
+ * Figure 6.19 - format_information field for AM824 Compound
+ * in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA)
+ * Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005
+ */
+int snd_oxfw_stream_parse_format(u8 *format,
+ struct snd_oxfw_stream_formation *formation)
+{
+ unsigned int i, e, channels, type;
+
+ memset(formation, 0, sizeof(struct snd_oxfw_stream_formation));
+
+ /*
+ * this module can support a hierarchy combination that:
+ * Root: Audio and Music (0x90)
+ * Level 1: AM824 Compound (0x40)
+ */
+ if ((format[0] != 0x90) || (format[1] != 0x40))
+ return -ENOSYS;
+
+ /* check the sampling rate */
+ for (i = 0; i < ARRAY_SIZE(avc_stream_rate_table); i++) {
+ if (format[2] == avc_stream_rate_table[i])
+ break;
+ }
+ if (i == ARRAY_SIZE(avc_stream_rate_table))
+ return -ENOSYS;
+
+ formation->rate = oxfw_rate_table[i];
+
+ for (e = 0; e < format[4]; e++) {
+ channels = format[5 + e * 2];
+ type = format[6 + e * 2];
+
+ switch (type) {
+ /* IEC 60958 Conformant, currently handled as MBLA */
+ case 0x00:
+ /* Multi Bit Linear Audio (Raw) */
+ case 0x06:
+ formation->pcm += channels;
+ break;
+ /* MIDI Conformant */
+ case 0x0d:
+ formation->midi = channels;
+ break;
+ /* IEC 61937-3 to 7 */
+ case 0x01:
+ case 0x02:
+ case 0x03:
+ case 0x04:
+ case 0x05:
+ /* Multi Bit Linear Audio */
+ case 0x07: /* DVD-Audio */
+ case 0x0c: /* High Precision */
+ /* One Bit Audio */
+ case 0x08: /* (Plain) Raw */
+ case 0x09: /* (Plain) SACD */
+ case 0x0a: /* (Encoded) Raw */
+ case 0x0b: /* (Encoded) SACD */
+ /* SMPTE Time-Code conformant */
+ case 0x0e:
+ /* Sample Count */
+ case 0x0f:
+ /* Anciliary Data */
+ case 0x10:
+ /* Synchronization Stream (Stereo Raw audio) */
+ case 0x40:
+ /* Don't care */
+ case 0xff:
+ default:
+ return -ENOSYS; /* not supported */
+ }
+ }
+
+ if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
+ formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ return -ENOSYS;
+
+ return 0;
+}
+
+static int
+assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
+ unsigned int pid, u8 *buf, unsigned int *len,
+ u8 **formats)
+{
+ struct snd_oxfw_stream_formation formation;
+ unsigned int i, eid;
+ int err;
+
+ /* get format at current sampling rate */
+ err = avc_stream_get_format_single(oxfw->unit, dir, pid, buf, len);
+ if (err < 0) {
+ dev_err(&oxfw->unit->device,
+ "fail to get current stream format for isoc %s plug %d:%d\n",
+ (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out",
+ pid, err);
+ goto end;
+ }
+
+ /* parse and set stream format */
+ eid = 0;
+ err = snd_oxfw_stream_parse_format(buf, &formation);
+ if (err < 0)
+ goto end;
+
+ formats[eid] = kmalloc(*len, GFP_KERNEL);
+ if (formats[eid] == NULL) {
+ err = -ENOMEM;
+ goto end;
+ }
+ memcpy(formats[eid], buf, *len);
+
+ /* apply the format for each available sampling rate */
+ for (i = 0; i < ARRAY_SIZE(oxfw_rate_table); i++) {
+ if (formation.rate == oxfw_rate_table[i])
+ continue;
+
+ err = avc_general_inquiry_sig_fmt(oxfw->unit,
+ oxfw_rate_table[i],
+ dir, pid);
+ if (err < 0)
+ continue;
+
+ eid++;
+ formats[eid] = kmalloc(*len, GFP_KERNEL);
+ if (formats[eid] == NULL) {
+ err = -ENOMEM;
+ goto end;
+ }
+ memcpy(formats[eid], buf, *len);
+ formats[eid][2] = avc_stream_rate_table[i];
+ }
+
+ err = 0;
+ oxfw->assumed = true;
+end:
+ return err;
+}
+
+static int fill_stream_formats(struct snd_oxfw *oxfw,
+ enum avc_general_plug_dir dir,
+ unsigned short pid)
+{
+ u8 *buf, **formats;
+ unsigned int len, eid = 0;
+ struct snd_oxfw_stream_formation dummy;
+ int err;
+
+ buf = kmalloc(AVC_GENERIC_FRAME_MAXIMUM_BYTES, GFP_KERNEL);
+ if (buf == NULL)
+ return -ENOMEM;
+
+ formats = oxfw->rx_stream_formats;
+
+ /* get first entry */
+ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
+ err = avc_stream_get_format_list(oxfw->unit, dir, 0, buf, &len, 0);
+ if (err == -ENOSYS) {
+ /* LIST subfunction is not implemented */
+ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
+ err = assume_stream_formats(oxfw, dir, pid, buf, &len,
+ formats);
+ goto end;
+ } else if (err < 0) {
+ dev_err(&oxfw->unit->device,
+ "fail to get stream format %d for isoc %s plug %d:%d\n",
+ eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out",
+ pid, err);
+ goto end;
+ }
+
+ /* LIST subfunction is implemented */
+ while (eid < SND_OXFW_STREAM_FORMAT_ENTRIES) {
+ /* The format is too short. */
+ if (len < 3) {
+ err = -EIO;
+ break;
+ }
+
+ /* parse and set stream format */
+ err = snd_oxfw_stream_parse_format(buf, &dummy);
+ if (err < 0)
+ break;
+
+ formats[eid] = kmalloc(len, GFP_KERNEL);
+ if (formats[eid] == NULL) {
+ err = -ENOMEM;
+ break;
+ }
+ memcpy(formats[eid], buf, len);
+
+ /* get next entry */
+ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
+ err = avc_stream_get_format_list(oxfw->unit, dir, 0,
+ buf, &len, ++eid);
+ /* No entries remained. */
+ if (err == -EINVAL) {
+ err = 0;
+ break;
+ } else if (err < 0) {
+ dev_err(&oxfw->unit->device,
+ "fail to get stream format %d for isoc %s plug %d:%d\n",
+ eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" :
+ "out",
+ pid, err);
+ break;
+ }
+ }
+end:
+ kfree(buf);
+ return err;
+}
+
+int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
+{
+ u8 plugs[AVC_PLUG_INFO_BUF_BYTES];
+ int err;
+
+ /* the number of plugs for isoc in/out, ext in/out */
+ err = avc_general_get_plug_info(oxfw->unit, 0x1f, 0x07, 0x00, plugs);
+ if (err < 0) {
+ dev_err(&oxfw->unit->device,
+ "fail to get info for isoc/external in/out plugs: %d\n",
+ err);
+ goto end;
+ } else if (plugs[0] == 0) {
+ err = -ENOSYS;
+ goto end;
+ }
+
+ /* use iPCR[0] if exists */
+ if (plugs[0] > 0)
+ err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
+end:
+ return err;
+}