[PATCH 22/29] ALSA: oxfw: Change the way to make PCM rules/constraints

[Takashi Sakamoto <o-takashi@sakamocchi.jp>]

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>
---
 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 1fd76ce..727d281 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 c262b56..f0b2c9c 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;
@@ -90,3 +119,242 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw)
 		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 08c55be..88ab913 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -25,6 +25,22 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@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;
@@ -128,24 +152,6 @@ 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);
-- 
1.9.1

_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel