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linux-next/sound/firewire/bebob/bebob_stream.c
Takashi Sakamoto 51fa31d462 ALSA: bebob: Improve comments about stream format
Currently bebob driver apply Raw Audio Data channel (in IEC 61883-1:2002,
Multi Bit Linear Audio Data channel in IEC 61883-6:20005) to IEC 60958
Conformant Data channel because both fireworks and bebob based devices
can handle it by ignoring each label.

This patch improves a comment about this.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-27 17:36:24 +02:00

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/*
* bebob_stream.c - a part of driver for BeBoB based devices
*
* Copyright (c) 2013-2014 Takashi Sakamoto
*
* Licensed under the terms of the GNU General Public License, version 2.
*/
#include "./bebob.h"
#define CALLBACK_TIMEOUT 1000
#define FW_ISO_RESOURCE_DELAY 1000
/*
* NOTE;
* For BeBoB streams, Both of input and output CMP connection are important.
*
* For most devices, each CMP connection starts to transmit/receive a
* corresponding stream. But for a few devices, both of CMP connection needs
* to start transmitting stream. An example is 'M-Audio Firewire 410'.
*/
/* 128 is an arbitrary length but it seems to be enough */
#define FORMAT_MAXIMUM_LENGTH 128
const unsigned int snd_bebob_rate_table[SND_BEBOB_STRM_FMT_ENTRIES] = {
[0] = 32000,
[1] = 44100,
[2] = 48000,
[3] = 88200,
[4] = 96000,
[5] = 176400,
[6] = 192000,
};
/*
* See: Table 51: Extended Stream Format Info Sampling Frequency
* in Additional AVC commands (Nov 2003, BridgeCo)
*/
static const unsigned int bridgeco_freq_table[] = {
[0] = 0x02,
[1] = 0x03,
[2] = 0x04,
[3] = 0x0a,
[4] = 0x05,
[5] = 0x06,
[6] = 0x07,
};
static unsigned int
get_formation_index(unsigned int rate)
{
unsigned int i;
for (i = 0; i < ARRAY_SIZE(snd_bebob_rate_table); i++) {
if (snd_bebob_rate_table[i] == rate)
return i;
}
return -EINVAL;
}
int
snd_bebob_stream_get_rate(struct snd_bebob *bebob, unsigned int *curr_rate)
{
unsigned int tx_rate, rx_rate, trials;
int err;
trials = 0;
do {
err = avc_general_get_sig_fmt(bebob->unit, &tx_rate,
AVC_GENERAL_PLUG_DIR_OUT, 0);
} while (err == -EAGAIN && ++trials < 3);
if (err < 0)
goto end;
trials = 0;
do {
err = avc_general_get_sig_fmt(bebob->unit, &rx_rate,
AVC_GENERAL_PLUG_DIR_IN, 0);
} while (err == -EAGAIN && ++trials < 3);
if (err < 0)
goto end;
*curr_rate = rx_rate;
if (rx_rate == tx_rate)
goto end;
/* synchronize receive stream rate to transmit stream rate */
err = avc_general_set_sig_fmt(bebob->unit, rx_rate,
AVC_GENERAL_PLUG_DIR_IN, 0);
end:
return err;
}
int
snd_bebob_stream_set_rate(struct snd_bebob *bebob, unsigned int rate)
{
int err;
err = avc_general_set_sig_fmt(bebob->unit, rate,
AVC_GENERAL_PLUG_DIR_OUT, 0);
if (err < 0)
goto end;
err = avc_general_set_sig_fmt(bebob->unit, rate,
AVC_GENERAL_PLUG_DIR_IN, 0);
if (err < 0)
goto end;
/*
* Some devices need a bit time for transition.
* 300msec is got by some experiments.
*/
msleep(300);
end:
return err;
}
int
snd_bebob_stream_check_internal_clock(struct snd_bebob *bebob, bool *internal)
{
struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
unsigned int id;
int err = 0;
*internal = false;
/* 1.The device has its own operation to switch source of clock */
if (clk_spec) {
err = clk_spec->get(bebob, &id);
if (err < 0)
dev_err(&bebob->unit->device,
"fail to get clock source: %d\n", err);
else if (strncmp(clk_spec->labels[id], SND_BEBOB_CLOCK_INTERNAL,
strlen(SND_BEBOB_CLOCK_INTERNAL)) == 0)
*internal = true;
goto end;
}
/*
* 2.The device don't support to switch source of clock then assumed
* to use internal clock always
*/
if (bebob->sync_input_plug < 0) {
*internal = true;
goto end;
}
/*
* 3.The device supports to switch source of clock by an usual way.
* Let's check input for 'Music Sub Unit Sync Input' plug.
*/
avc_bridgeco_fill_msu_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN,
bebob->sync_input_plug);
err = avc_bridgeco_get_plug_input(bebob->unit, addr, input);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get an input for MSU in plug %d: %d\n",
bebob->sync_input_plug, err);
goto end;
}
/*
* If there are no input plugs, all of fields are 0xff.
* Here check the first field. This field is used for direction.
*/
if (input[0] == 0xff) {
*internal = true;
goto end;
}
/*
* If source of clock is internal CSR, Music Sub Unit Sync Input is
* a destination of Music Sub Unit Sync Output.
*/
*internal = ((input[0] == AVC_BRIDGECO_PLUG_DIR_OUT) &&
(input[1] == AVC_BRIDGECO_PLUG_MODE_SUBUNIT) &&
(input[2] == 0x0c) &&
(input[3] == 0x00));
end:
return err;
}
static unsigned int
map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
{
unsigned int sec, sections, ch, channels;
unsigned int pcm, midi, location;
unsigned int stm_pos, sec_loc, pos;
u8 *buf, addr[AVC_BRIDGECO_ADDR_BYTES], type;
enum avc_bridgeco_plug_dir dir;
int err;
/*
* The length of return value of this command cannot be expected. Here
* use the maximum length of FCP.
*/
buf = kzalloc(256, GFP_KERNEL);
if (buf == NULL)
return -ENOMEM;
if (s == &bebob->tx_stream)
dir = AVC_BRIDGECO_PLUG_DIR_OUT;
else
dir = AVC_BRIDGECO_PLUG_DIR_IN;
avc_bridgeco_fill_unit_addr(addr, dir, AVC_BRIDGECO_PLUG_UNIT_ISOC, 0);
err = avc_bridgeco_get_plug_ch_pos(bebob->unit, addr, buf, 256);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get channel position for isoc %s plug 0: %d\n",
(dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" : "out",
err);
goto end;
}
pos = 0;
/* positions in I/O buffer */
pcm = 0;
midi = 0;
/* the number of sections in AMDTP packet */
sections = buf[pos++];
for (sec = 0; sec < sections; sec++) {
/* type of this section */
avc_bridgeco_fill_unit_addr(addr, dir,
AVC_BRIDGECO_PLUG_UNIT_ISOC, 0);
err = avc_bridgeco_get_plug_section_type(bebob->unit, addr,
sec, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get section type for isoc %s plug 0: %d\n",
(dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" :
"out",
err);
goto end;
}
/* NoType */
if (type == 0xff) {
err = -ENOSYS;
goto end;
}
/* the number of channels in this section */
channels = buf[pos++];
for (ch = 0; ch < channels; ch++) {
/* position of this channel in AMDTP packet */
stm_pos = buf[pos++] - 1;
/* location of this channel in this section */
sec_loc = buf[pos++] - 1;
/*
* Basically the number of location is within the
* number of channels in this section. But some models
* of M-Audio don't follow this. Its location for MIDI
* is the position of MIDI channels in AMDTP packet.
*/
if (sec_loc >= channels)
sec_loc = ch;
switch (type) {
/* for MIDI conformant data channel */
case 0x0a:
/* AMDTP_MAX_CHANNELS_FOR_MIDI is 1. */
if ((midi > 0) && (stm_pos != midi)) {
err = -ENOSYS;
goto end;
}
s->midi_position = stm_pos;
midi = stm_pos;
break;
/* for PCM data channel */
case 0x01: /* Headphone */
case 0x02: /* Microphone */
case 0x03: /* Line */
case 0x04: /* SPDIF */
case 0x05: /* ADAT */
case 0x06: /* TDIF */
case 0x07: /* MADI */
/* for undefined/changeable signal */
case 0x08: /* Analog */
case 0x09: /* Digital */
default:
location = pcm + sec_loc;
if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
s->pcm_positions[location] = stm_pos;
break;
}
}
if (type != 0x0a)
pcm += channels;
else
midi += channels;
}
end:
kfree(buf);
return err;
}
static int
init_both_connections(struct snd_bebob *bebob)
{
int err;
err = cmp_connection_init(&bebob->in_conn,
bebob->unit, CMP_INPUT, 0);
if (err < 0)
goto end;
err = cmp_connection_init(&bebob->out_conn,
bebob->unit, CMP_OUTPUT, 0);
if (err < 0)
cmp_connection_destroy(&bebob->in_conn);
end:
return err;
}
static int
check_connection_used_by_others(struct snd_bebob *bebob, struct amdtp_stream *s)
{
struct cmp_connection *conn;
bool used;
int err;
if (s == &bebob->tx_stream)
conn = &bebob->out_conn;
else
conn = &bebob->in_conn;
err = cmp_connection_check_used(conn, &used);
if ((err >= 0) && used && !amdtp_stream_running(s)) {
dev_err(&bebob->unit->device,
"Connection established by others: %cPCR[%d]\n",
(conn->direction == CMP_OUTPUT) ? 'o' : 'i',
conn->pcr_index);
err = -EBUSY;
}
return err;
}
static int
make_both_connections(struct snd_bebob *bebob, unsigned int rate)
{
int index, pcm_channels, midi_channels, err = 0;
if (bebob->connected)
goto end;
/* confirm params for both streams */
index = get_formation_index(rate);
pcm_channels = bebob->tx_stream_formations[index].pcm;
midi_channels = bebob->tx_stream_formations[index].midi;
amdtp_stream_set_parameters(&bebob->tx_stream,
rate, pcm_channels, midi_channels * 8);
pcm_channels = bebob->rx_stream_formations[index].pcm;
midi_channels = bebob->rx_stream_formations[index].midi;
amdtp_stream_set_parameters(&bebob->rx_stream,
rate, pcm_channels, midi_channels * 8);
/* establish connections for both streams */
err = cmp_connection_establish(&bebob->out_conn,
amdtp_stream_get_max_payload(&bebob->tx_stream));
if (err < 0)
goto end;
err = cmp_connection_establish(&bebob->in_conn,
amdtp_stream_get_max_payload(&bebob->rx_stream));
if (err < 0) {
cmp_connection_break(&bebob->out_conn);
goto end;
}
bebob->connected = true;
end:
return err;
}
static void
break_both_connections(struct snd_bebob *bebob)
{
cmp_connection_break(&bebob->in_conn);
cmp_connection_break(&bebob->out_conn);
bebob->connected = false;
/* These models seems to be in transition state for a longer time. */
if (bebob->maudio_special_quirk != NULL)
msleep(200);
}
static void
destroy_both_connections(struct snd_bebob *bebob)
{
break_both_connections(bebob);
cmp_connection_destroy(&bebob->in_conn);
cmp_connection_destroy(&bebob->out_conn);
}
static int
get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode)
{
/* currently this module doesn't support SYT-Match mode */
*sync_mode = CIP_SYNC_TO_DEVICE;
return 0;
}
static int
start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream,
unsigned int rate)
{
struct cmp_connection *conn;
int err = 0;
if (stream == &bebob->rx_stream)
conn = &bebob->in_conn;
else
conn = &bebob->out_conn;
/* channel mapping */
if (bebob->maudio_special_quirk == NULL) {
err = map_data_channels(bebob, stream);
if (err < 0)
goto end;
}
/* start amdtp stream */
err = amdtp_stream_start(stream,
conn->resources.channel,
conn->speed);
end:
return err;
}
int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
{
int err;
err = init_both_connections(bebob);
if (err < 0)
goto end;
err = amdtp_stream_init(&bebob->tx_stream, bebob->unit,
AMDTP_IN_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
goto end;
}
/* See comments in next function */
init_completion(&bebob->bus_reset);
bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK;
/*
* At high sampling rate, M-Audio special firmware transmits empty
* packet with the value of dbc incremented by 8 but the others are
* valid to IEC 61883-1.
*/
if (bebob->maudio_special_quirk)
bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC;
err = amdtp_stream_init(&bebob->rx_stream, bebob->unit,
AMDTP_OUT_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
amdtp_stream_destroy(&bebob->rx_stream);
destroy_both_connections(bebob);
}
/*
* The firmware for these devices ignore MIDI messages in more than
* first 8 data blocks of an received AMDTP packet.
*/
if (bebob->spec == &maudio_fw410_spec ||
bebob->spec == &maudio_special_spec)
bebob->rx_stream.rx_blocks_for_midi = 8;
end:
return err;
}
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
{
struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
struct amdtp_stream *master, *slave;
atomic_t *slave_substreams;
enum cip_flags sync_mode;
unsigned int curr_rate;
bool updated = false;
int err = 0;
/*
* Normal BeBoB firmware has a quirk at bus reset to transmits packets
* with discontinuous value in dbc field.
*
* This 'struct completion' is used to call .update() at first to update
* connections/streams. Next following codes handle streaming error.
*/
if (amdtp_streaming_error(&bebob->tx_stream)) {
if (completion_done(&bebob->bus_reset))
reinit_completion(&bebob->bus_reset);
updated = (wait_for_completion_interruptible_timeout(
&bebob->bus_reset,
msecs_to_jiffies(FW_ISO_RESOURCE_DELAY)) > 0);
}
mutex_lock(&bebob->mutex);
/* Need no substreams */
if (atomic_read(&bebob->playback_substreams) == 0 &&
atomic_read(&bebob->capture_substreams) == 0)
goto end;
err = get_sync_mode(bebob, &sync_mode);
if (err < 0)
goto end;
if (sync_mode == CIP_SYNC_TO_DEVICE) {
master = &bebob->tx_stream;
slave = &bebob->rx_stream;
slave_substreams = &bebob->playback_substreams;
} else {
master = &bebob->rx_stream;
slave = &bebob->tx_stream;
slave_substreams = &bebob->capture_substreams;
}
/*
* Considering JACK/FFADO streaming:
* TODO: This can be removed hwdep functionality becomes popular.
*/
err = check_connection_used_by_others(bebob, master);
if (err < 0)
goto end;
/*
* packet queueing error or detecting discontinuity
*
* At bus reset, connections should not be broken here. So streams need
* to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag.
*/
if (amdtp_streaming_error(master))
amdtp_stream_stop(master);
if (amdtp_streaming_error(slave))
amdtp_stream_stop(slave);
if (!updated &&
!amdtp_stream_running(master) && !amdtp_stream_running(slave))
break_both_connections(bebob);
/* stop streams if rate is different */
err = rate_spec->get(bebob, &curr_rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get sampling rate: %d\n", err);
goto end;
}
if (rate == 0)
rate = curr_rate;
if (rate != curr_rate) {
amdtp_stream_stop(master);
amdtp_stream_stop(slave);
break_both_connections(bebob);
}
/* master should be always running */
if (!amdtp_stream_running(master)) {
amdtp_stream_set_sync(sync_mode, master, slave);
bebob->master = master;
/*
* NOTE:
* If establishing connections at first, Yamaha GO46
* (and maybe Terratec X24) don't generate sound.
*
* For firmware customized by M-Audio, refer to next NOTE.
*/
if (bebob->maudio_special_quirk == NULL) {
err = rate_spec->set(bebob, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to set sampling rate: %d\n",
err);
goto end;
}
}
err = make_both_connections(bebob, rate);
if (err < 0)
goto end;
err = start_stream(bebob, master, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to run AMDTP master stream:%d\n", err);
break_both_connections(bebob);
goto end;
}
/*
* NOTE:
* The firmware customized by M-Audio uses these commands to
* start transmitting stream. This is not usual way.
*/
if (bebob->maudio_special_quirk != NULL) {
err = rate_spec->set(bebob, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to ensure sampling rate: %d\n",
err);
amdtp_stream_stop(master);
break_both_connections(bebob);
goto end;
}
}
/* wait first callback */
if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) {
amdtp_stream_stop(master);
break_both_connections(bebob);
err = -ETIMEDOUT;
goto end;
}
}
/* start slave if needed */
if (atomic_read(slave_substreams) > 0 && !amdtp_stream_running(slave)) {
err = start_stream(bebob, slave, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to run AMDTP slave stream:%d\n", err);
amdtp_stream_stop(master);
break_both_connections(bebob);
goto end;
}
/* wait first callback */
if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) {
amdtp_stream_stop(slave);
amdtp_stream_stop(master);
break_both_connections(bebob);
err = -ETIMEDOUT;
}
}
end:
mutex_unlock(&bebob->mutex);
return err;
}
void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
{
struct amdtp_stream *master, *slave;
atomic_t *master_substreams, *slave_substreams;
mutex_lock(&bebob->mutex);
if (bebob->master == &bebob->rx_stream) {
slave = &bebob->tx_stream;
master = &bebob->rx_stream;
slave_substreams = &bebob->capture_substreams;
master_substreams = &bebob->playback_substreams;
} else {
slave = &bebob->rx_stream;
master = &bebob->tx_stream;
slave_substreams = &bebob->playback_substreams;
master_substreams = &bebob->capture_substreams;
}
if (atomic_read(slave_substreams) == 0) {
amdtp_stream_pcm_abort(slave);
amdtp_stream_stop(slave);
if (atomic_read(master_substreams) == 0) {
amdtp_stream_pcm_abort(master);
amdtp_stream_stop(master);
break_both_connections(bebob);
}
}
mutex_unlock(&bebob->mutex);
}
void snd_bebob_stream_update_duplex(struct snd_bebob *bebob)
{
/* vs. XRUN recovery due to discontinuity at bus reset */
mutex_lock(&bebob->mutex);
if ((cmp_connection_update(&bebob->in_conn) < 0) ||
(cmp_connection_update(&bebob->out_conn) < 0)) {
amdtp_stream_pcm_abort(&bebob->rx_stream);
amdtp_stream_pcm_abort(&bebob->tx_stream);
amdtp_stream_stop(&bebob->rx_stream);
amdtp_stream_stop(&bebob->tx_stream);
break_both_connections(bebob);
} else {
amdtp_stream_update(&bebob->rx_stream);
amdtp_stream_update(&bebob->tx_stream);
}
/* wake up stream_start_duplex() */
if (!completion_done(&bebob->bus_reset))
complete_all(&bebob->bus_reset);
mutex_unlock(&bebob->mutex);
}
void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
{
mutex_lock(&bebob->mutex);
amdtp_stream_pcm_abort(&bebob->rx_stream);
amdtp_stream_pcm_abort(&bebob->tx_stream);
amdtp_stream_stop(&bebob->rx_stream);
amdtp_stream_stop(&bebob->tx_stream);
amdtp_stream_destroy(&bebob->rx_stream);
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
mutex_unlock(&bebob->mutex);
}
/*
* See: Table 50: Extended Stream Format Info Format Hierarchy Level 2
* in Additional AVC commands (Nov 2003, BridgeCo)
* Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005
*/
static int
parse_stream_formation(u8 *buf, unsigned int len,
struct snd_bebob_stream_formation *formation)
{
unsigned int i, e, channels, format;
/*
* this module can support a hierarchy combination that:
* Root: Audio and Music (0x90)
* Level 1: AM824 Compound (0x40)
*/
if ((buf[0] != 0x90) || (buf[1] != 0x40))
return -ENOSYS;
/* check sampling rate */
for (i = 0; i < ARRAY_SIZE(bridgeco_freq_table); i++) {
if (buf[2] == bridgeco_freq_table[i])
break;
}
if (i == sizeof(bridgeco_freq_table))
return -ENOSYS;
/* Avoid double count by different entries for the same rate. */
memset(&formation[i], 0, sizeof(struct snd_bebob_stream_formation));
for (e = 0; e < buf[4]; e++) {
channels = buf[5 + e * 2];
format = buf[6 + e * 2];
switch (format) {
/* IEC 60958 Conformant, currently handled as MBLA */
case 0x00:
/* Multi bit linear audio */
case 0x06: /* Raw */
formation[i].pcm += channels;
break;
/* MIDI Conformant */
case 0x0d:
formation[i].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 */
/* Synchronization Stream (Stereo Raw audio) */
case 0x40:
/* Don't care */
case 0xff:
default:
return -ENOSYS; /* not supported */
}
}
if (formation[i].pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
}
static int
fill_stream_formations(struct snd_bebob *bebob, enum avc_bridgeco_plug_dir dir,
unsigned short pid)
{
u8 *buf;
struct snd_bebob_stream_formation *formations;
unsigned int len, eid;
u8 addr[AVC_BRIDGECO_ADDR_BYTES];
int err;
buf = kmalloc(FORMAT_MAXIMUM_LENGTH, GFP_KERNEL);
if (buf == NULL)
return -ENOMEM;
if (dir == AVC_BRIDGECO_PLUG_DIR_IN)
formations = bebob->rx_stream_formations;
else
formations = bebob->tx_stream_formations;
for (eid = 0; eid < SND_BEBOB_STRM_FMT_ENTRIES; eid++) {
len = FORMAT_MAXIMUM_LENGTH;
avc_bridgeco_fill_unit_addr(addr, dir,
AVC_BRIDGECO_PLUG_UNIT_ISOC, pid);
err = avc_bridgeco_get_plug_strm_fmt(bebob->unit, addr, buf,
&len, eid);
/* No entries remained. */
if (err == -EINVAL && eid > 0) {
err = 0;
break;
} else if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get stream format %d for isoc %s plug %d:%d\n",
eid,
(dir == AVC_BRIDGECO_PLUG_DIR_IN) ? "in" :
"out",
pid, err);
break;
}
err = parse_stream_formation(buf, len, formations);
if (err < 0)
break;
}
kfree(buf);
return err;
}
static int
seek_msu_sync_input_plug(struct snd_bebob *bebob)
{
u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
unsigned int i;
enum avc_bridgeco_plug_type type;
int err;
/* Get the number of Music Sub Unit for both direction. */
err = avc_general_get_plug_info(bebob->unit, 0x0c, 0x00, 0x00, plugs);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get info for MSU in/out plugs: %d\n",
err);
goto end;
}
/* seek destination plugs for 'MSU sync input' */
bebob->sync_input_plug = -1;
for (i = 0; i < plugs[0]; i++) {
avc_bridgeco_fill_msu_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN, i);
err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get type for MSU in plug %d: %d\n",
i, err);
goto end;
}
if (type == AVC_BRIDGECO_PLUG_TYPE_SYNC) {
bebob->sync_input_plug = i;
break;
}
}
end:
return err;
}
int snd_bebob_stream_discover(struct snd_bebob *bebob)
{
struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
enum avc_bridgeco_plug_type type;
unsigned int i;
int err;
/* the number of plugs for isoc in/out, ext in/out */
err = avc_general_get_plug_info(bebob->unit, 0x1f, 0x07, 0x00, plugs);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get info for isoc/external in/out plugs: %d\n",
err);
goto end;
}
/*
* This module supports at least one isoc input plug and one isoc
* output plug.
*/
if ((plugs[0] == 0) || (plugs[1] == 0)) {
err = -ENOSYS;
goto end;
}
avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN,
AVC_BRIDGECO_PLUG_UNIT_ISOC, 0);
err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get type for isoc in plug 0: %d\n", err);
goto end;
} else if (type != AVC_BRIDGECO_PLUG_TYPE_ISOC) {
err = -ENOSYS;
goto end;
}
err = fill_stream_formations(bebob, AVC_BRIDGECO_PLUG_DIR_IN, 0);
if (err < 0)
goto end;
avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_OUT,
AVC_BRIDGECO_PLUG_UNIT_ISOC, 0);
err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get type for isoc out plug 0: %d\n", err);
goto end;
} else if (type != AVC_BRIDGECO_PLUG_TYPE_ISOC) {
err = -ENOSYS;
goto end;
}
err = fill_stream_formations(bebob, AVC_BRIDGECO_PLUG_DIR_OUT, 0);
if (err < 0)
goto end;
/* count external input plugs for MIDI */
bebob->midi_input_ports = 0;
for (i = 0; i < plugs[2]; i++) {
avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN,
AVC_BRIDGECO_PLUG_UNIT_EXT, i);
err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get type for external in plug %d: %d\n",
i, err);
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_MIDI) {
bebob->midi_input_ports++;
}
}
/* count external output plugs for MIDI */
bebob->midi_output_ports = 0;
for (i = 0; i < plugs[3]; i++) {
avc_bridgeco_fill_unit_addr(addr, AVC_BRIDGECO_PLUG_DIR_OUT,
AVC_BRIDGECO_PLUG_UNIT_EXT, i);
err = avc_bridgeco_get_plug_type(bebob->unit, addr, &type);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to get type for external out plug %d: %d\n",
i, err);
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_MIDI) {
bebob->midi_output_ports++;
}
}
/* for check source of clock later */
if (!clk_spec)
err = seek_msu_sync_input_plug(bebob);
end:
return err;
}
void snd_bebob_stream_lock_changed(struct snd_bebob *bebob)
{
bebob->dev_lock_changed = true;
wake_up(&bebob->hwdep_wait);
}
int snd_bebob_stream_lock_try(struct snd_bebob *bebob)
{
int err;
spin_lock_irq(&bebob->lock);
/* user land lock this */
if (bebob->dev_lock_count < 0) {
err = -EBUSY;
goto end;
}
/* this is the first time */
if (bebob->dev_lock_count++ == 0)
snd_bebob_stream_lock_changed(bebob);
err = 0;
end:
spin_unlock_irq(&bebob->lock);
return err;
}
void snd_bebob_stream_lock_release(struct snd_bebob *bebob)
{
spin_lock_irq(&bebob->lock);
if (WARN_ON(bebob->dev_lock_count <= 0))
goto end;
if (--bebob->dev_lock_count == 0)
snd_bebob_stream_lock_changed(bebob);
end:
spin_unlock_irq(&bebob->lock);
}