[alsa-devel] [PATCH 23/30] ALSA: oxfw: Change the way to make PCM rules/constraints
Takashi Sakamoto
o-takashi at sakamocchi.jp
Fri Nov 28 16:59:32 CET 2014
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 at sakamocchi.jp>
---
sound/firewire/oxfw/oxfw-pcm.c | 197 ++++++++++++++++------------
sound/firewire/oxfw/oxfw-stream.c | 268 ++++++++++++++++++++++++++++++++++++++
sound/firewire/oxfw/oxfw.c | 42 +++---
sound/firewire/oxfw/oxfw.h | 21 ++-
4 files changed, 426 insertions(+), 102 deletions(-)
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index d39f17a..0c0be98 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -7,117 +7,152 @@
#include "oxfw.h"
-static int firewave_rate_constraint(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
{
- static unsigned int stereo_rates[] = { 48000, 96000 };
- struct snd_interval *channels =
- hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
- struct snd_interval *rate =
- hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
-
- /* two channels work only at 48/96 kHz */
- if (snd_interval_max(channels) < 6)
- return snd_interval_list(rate, 2, stereo_rates, 0);
- return 0;
+ u8 **formats = rule->private;
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1
+ };
+ struct snd_oxfw_stream_formation formation;
+ unsigned int i, err;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ if (formats[i] == NULL)
+ continue;
+
+ err = snd_oxfw_stream_parse_format(formats[i], &formation);
+ if (err < 0)
+ continue;
+ if (!snd_interval_test(c, formation.pcm))
+ continue;
+
+ t.min = min(t.min, formation.rate);
+ t.max = max(t.max, formation.rate);
+
+ }
+ return snd_interval_refine(r, &t);
}
-static int firewave_channels_constraint(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
{
- static const struct snd_interval all_channels = { .min = 6, .max = 6 };
- struct snd_interval *rate =
- hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
- struct snd_interval *channels =
- hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
-
- /* 32/44.1 kHz work only with all six channels */
- if (snd_interval_max(rate) < 48000)
- return snd_interval_refine(channels, &all_channels);
- return 0;
+ u8 **formats = rule->private;
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_oxfw_stream_formation formation;
+ unsigned int i, j, err;
+ unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0};
+
+ count = 0;
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ if (formats[i] == NULL)
+ break;
+
+ err = snd_oxfw_stream_parse_format(formats[i], &formation);
+ if (err < 0)
+ continue;
+ if (!snd_interval_test(r, formation.rate))
+ continue;
+ if (list[count] == formation.pcm)
+ continue;
+
+ for (j = 0; j < ARRAY_SIZE(list); j++) {
+ if (list[j] == formation.pcm)
+ break;
+ }
+ if (j == ARRAY_SIZE(list)) {
+ list[count] = formation.pcm;
+ if (++count == ARRAY_SIZE(list))
+ break;
+ }
+ }
+
+ return snd_interval_list(c, count, list, 0);
}
-int firewave_constraints(struct snd_pcm_runtime *runtime)
+static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats)
{
- static unsigned int channels_list[] = { 2, 6 };
- static struct snd_pcm_hw_constraint_list channels_list_constraint = {
- .count = 2,
- .list = channels_list,
- };
- int err;
+ struct snd_oxfw_stream_formation formation;
+ unsigned int i, err;
- runtime->hw.rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 |
- SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_96000;
- runtime->hw.channels_max = 6;
+ hw->channels_min = UINT_MAX;
+ hw->channels_max = 0;
- err = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- &channels_list_constraint);
- if (err < 0)
- return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- firewave_rate_constraint, NULL,
- SNDRV_PCM_HW_PARAM_CHANNELS, -1);
- if (err < 0)
- return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- firewave_channels_constraint, NULL,
- SNDRV_PCM_HW_PARAM_RATE, -1);
- if (err < 0)
- return err;
+ hw->rate_min = UINT_MAX;
+ hw->rate_max = 0;
+ hw->rates = 0;
- return 0;
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ if (formats[i] == NULL)
+ break;
+
+ err = snd_oxfw_stream_parse_format(formats[i], &formation);
+ if (err < 0)
+ continue;
+
+ hw->channels_min = min(hw->channels_min, formation.pcm);
+ hw->channels_max = max(hw->channels_max, formation.pcm);
+
+ hw->rate_min = min(hw->rate_min, formation.rate);
+ hw->rate_max = max(hw->rate_max, formation.rate);
+ hw->rates |= snd_pcm_rate_to_rate_bit(formation.rate);
+ }
}
-int lacie_speakers_constraints(struct snd_pcm_runtime *runtime)
+static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
{
- runtime->hw.rates = SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 |
- SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 |
- SNDRV_PCM_RATE_96000;
+ hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */
+ hw->periods_max = UINT_MAX;
- return 0;
+ hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */
+
+ /* Just to prevent from allocating much pages. */
+ hw->period_bytes_max = hw->period_bytes_min * 2048;
+ hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
}
static int pcm_open(struct snd_pcm_substream *substream)
{
- static const struct snd_pcm_hardware hardware = {
- .info = SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_BATCH |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER,
- .formats = AMDTP_OUT_PCM_FORMAT_BITS,
- .channels_min = 2,
- .channels_max = 2,
- .buffer_bytes_max = 4 * 1024 * 1024,
- .period_bytes_min = 1,
- .period_bytes_max = UINT_MAX,
- .periods_min = 1,
- .periods_max = UINT_MAX,
- };
struct snd_oxfw *oxfw = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
- bool used;
+ u8 **formats;
int err;
- err = cmp_connection_check_used(&oxfw->in_conn, &used);
- if ((err < 0) || used)
- goto end;
+ formats = oxfw->rx_stream_formats;
+
+ runtime->hw.info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID;
- runtime->hw = hardware;
+ limit_channels_and_rates(&runtime->hw, formats);
+ limit_period_and_buffer(&runtime->hw);
- err = oxfw->device_info->pcm_constraints(runtime);
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, formats,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
if (err < 0)
goto end;
- err = snd_pcm_limit_hw_rates(runtime);
+
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, formats,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
if (err < 0)
goto end;
err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime);
+ if (err < 0)
+ goto end;
+
+ snd_pcm_set_sync(substream);
end:
return err;
}
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index ebd156f..17e3802 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;
+}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 3308253..4514650 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -25,6 +25,22 @@ MODULE_AUTHOR("Clemens Ladisch <clemens at ladisch.de>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("snd-firewire-speakers");
+static const struct device_info griffin_firewave = {
+ .vendor_name = "Griffin",
+ .driver_name = "FireWave",
+ .mixer_channels = 6,
+ .mute_fb_id = 0x01,
+ .volume_fb_id = 0x02,
+};
+
+static const struct device_info lacie_speakers = {
+ .vendor_name = "LaCie",
+ .driver_name = "FWSpeakers",
+ .mixer_channels = 1,
+ .mute_fb_id = 0x01,
+ .volume_fb_id = 0x01,
+};
+
static int name_card(struct snd_oxfw *oxfw)
{
struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
@@ -58,6 +74,10 @@ end:
static void oxfw_card_free(struct snd_card *card)
{
struct snd_oxfw *oxfw = card->private_data;
+ unsigned int i;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++)
+ kfree(oxfw->rx_stream_formats[i]);
mutex_destroy(&oxfw->mutex);
}
@@ -81,6 +101,10 @@ static int oxfw_probe(struct fw_unit *unit,
oxfw->unit = unit;
oxfw->device_info = (const struct device_info *)id->driver_data;
+ err = snd_oxfw_stream_discover(oxfw);
+ if (err < 0)
+ goto error;
+
err = name_card(oxfw);
if (err < 0)
goto error;
@@ -132,24 +156,6 @@ static void oxfw_remove(struct fw_unit *unit)
snd_card_free_when_closed(oxfw->card);
}
-static const struct device_info griffin_firewave = {
- .vendor_name = "Griffin",
- .driver_name = "FireWave",
- .pcm_constraints = firewave_constraints,
- .mixer_channels = 6,
- .mute_fb_id = 0x01,
- .volume_fb_id = 0x02,
-};
-
-static const struct device_info lacie_speakers = {
- .vendor_name = "LaCie",
- .driver_name = "FWSpeakers",
- .pcm_constraints = lacie_speakers_constraints,
- .mixer_channels = 1,
- .mute_fb_id = 0x01,
- .volume_fb_id = 0x01,
-};
-
static const struct ieee1394_device_id oxfw_id_table[] = {
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index 583ee7b..c4c1acd 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -29,19 +29,24 @@
struct device_info {
const char *vendor_name;
const char *driver_name;
- int (*pcm_constraints)(struct snd_pcm_runtime *runtime);
unsigned int mixer_channels;
u8 mute_fb_id;
u8 volume_fb_id;
};
+/* This is an arbitrary number for convinience. */
+#define SND_OXFW_STREAM_FORMAT_ENTRIES 10
struct snd_oxfw {
struct snd_card *card;
struct fw_unit *unit;
const struct device_info *device_info;
struct mutex mutex;
+
+ u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
+ bool assumed;
struct cmp_connection in_conn;
struct amdtp_stream rx_stream;
+
bool mute;
s16 volume[6];
s16 volume_min;
@@ -87,8 +92,18 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw);
void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw);
void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw);
-int firewave_constraints(struct snd_pcm_runtime *runtime);
-int lacie_speakers_constraints(struct snd_pcm_runtime *runtime);
+struct snd_oxfw_stream_formation {
+ unsigned int rate;
+ unsigned int pcm;
+ unsigned int midi;
+};
+int snd_oxfw_stream_parse_format(u8 *format,
+ struct snd_oxfw_stream_formation *formation);
+int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw,
+ enum avc_general_plug_dir dir,
+ struct snd_oxfw_stream_formation *formation);
+int snd_oxfw_stream_discover(struct snd_oxfw *oxfw);
+
int snd_oxfw_create_pcm(struct snd_oxfw *oxfw);
int snd_oxfw_create_mixer(struct snd_oxfw *oxfw);
--
2.1.0
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