Commit 5cd1d3f4 authored by Takashi Sakamoto's avatar Takashi Sakamoto Committed by Takashi Iwai

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: default avatarTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: default avatarClemens Ladisch <clemens@ladisch.de>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent 5b59d809
......@@ -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;
}
......
......@@ -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;
}
......@@ -58,6 +58,10 @@ static int name_card(struct snd_oxfw *oxfw)
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 +85,10 @@ 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;
......@@ -136,7 +144,6 @@ static const struct device_info griffin_firewave = {
.driver_name = "FireWave",
.vendor_name = "Griffin",
.model_name = "FireWave",
.pcm_constraints = firewave_constraints,
.mixer_channels = 6,
.mute_fb_id = 0x01,
.volume_fb_id = 0x02,
......@@ -146,7 +153,6 @@ static const struct device_info lacie_speakers = {
.driver_name = "FWSpeakers",
.vendor_name = "LaCie",
.model_name = "FireWire Speakers",
.pcm_constraints = lacie_speakers_constraints,
.mixer_channels = 1,
.mute_fb_id = 0x01,
.volume_fb_id = 0x01,
......
......@@ -30,19 +30,24 @@ struct device_info {
const char *driver_name;
const char *vendor_name;
const char *model_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;
......@@ -88,8 +93,18 @@ 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);
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