linux_dsm_epyc7002/sound/firewire/bebob/bebob_stream.c
Takashi Sakamoto 62f00e40b0 ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag
In former commit, drivers in ALSA firewire stack always starts IT context
before IR context. If IR context starts after packets are transmitted by
peer unit, packet discontinuity may be detected because the context starts
in the middle of packet streaming. This situation is rare because IT
context usually starts immediately. However, it's better to solve this
issue. This is suppressed with CIP_SKIP_INIT_DBC_CHECK flag.

This commit enables the same feature as CIP_SKIP_INIT_DBC_CHECK.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-05-10 17:04:01 +02:00

1006 lines
24 KiB
C
Raw Blame History

This file contains ambiguous Unicode characters

This file contains Unicode characters that might be confused with other characters. If you think that this is intentional, you can safely ignore this warning. Use the Escape button to reveal them.

/*
* 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 2000
#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 int
get_formation_index(unsigned int rate, unsigned int *index)
{
unsigned int i;
for (i = 0; i < ARRAY_SIZE(snd_bebob_rate_table); i++) {
if (snd_bebob_rate_table[i] == rate) {
*index = i;
return 0;
}
}
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_get_clock_src(struct snd_bebob *bebob,
enum snd_bebob_clock_type *src)
{
const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
unsigned int id;
enum avc_bridgeco_plug_type type;
int err = 0;
/* 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);
goto end;
}
if (id >= clk_spec->num) {
dev_err(&bebob->unit->device,
"clock source %d out of range 0..%d\n",
id, clk_spec->num - 1);
err = -EIO;
goto end;
}
*src = clk_spec->types[id];
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) {
*src = SND_BEBOB_CLOCK_TYPE_INTERNAL;
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) {
*src = SND_BEBOB_CLOCK_TYPE_INTERNAL;
goto end;
}
/* The source from any output plugs is for one purpose only. */
if (input[0] == AVC_BRIDGECO_PLUG_DIR_OUT) {
/*
* In BeBoB architecture, the source from music subunit may
* bypass from oPCR[0]. This means that this source gives
* synchronization to IEEE 1394 cycle start packet.
*/
if (input[1] == AVC_BRIDGECO_PLUG_MODE_SUBUNIT &&
input[2] == 0x0c) {
*src = SND_BEBOB_CLOCK_TYPE_INTERNAL;
goto end;
}
/* The source from any input units is for several purposes. */
} else if (input[1] == AVC_BRIDGECO_PLUG_MODE_UNIT) {
if (input[2] == AVC_BRIDGECO_PLUG_UNIT_ISOC) {
if (input[3] == 0x00) {
/*
* This source comes from iPCR[0]. This means
* that presentation timestamp calculated by
* SYT series of the received packets. In
* short, this driver is the master of
* synchronization.
*/
*src = SND_BEBOB_CLOCK_TYPE_SYT;
goto end;
} else {
/*
* This source comes from iPCR[1-29]. This
* means that the synchronization stream is not
* the Audio/MIDI compound stream.
*/
*src = SND_BEBOB_CLOCK_TYPE_EXTERNAL;
goto end;
}
} else if (input[2] == AVC_BRIDGECO_PLUG_UNIT_EXT) {
/* Check type of this plug. */
avc_bridgeco_fill_unit_addr(addr,
AVC_BRIDGECO_PLUG_DIR_IN,
AVC_BRIDGECO_PLUG_UNIT_EXT,
input[3]);
err = avc_bridgeco_get_plug_type(bebob->unit, addr,
&type);
if (err < 0)
goto end;
if (type == AVC_BRIDGECO_PLUG_TYPE_DIG) {
/*
* SPDIF/ADAT or sometimes (not always) word
* clock.
*/
*src = SND_BEBOB_CLOCK_TYPE_EXTERNAL;
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_SYNC) {
/* Often word clock. */
*src = SND_BEBOB_CLOCK_TYPE_EXTERNAL;
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_ADDITION) {
/*
* Not standard.
* Mostly, additional internal clock.
*/
*src = SND_BEBOB_CLOCK_TYPE_INTERNAL;
goto end;
}
}
}
/* Not supported. */
err = -EIO;
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;
}
amdtp_am824_set_midi_position(s, 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 >= AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
amdtp_am824_set_pcm_position(s, 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 */
err = get_formation_index(rate, &index);
if (err < 0)
goto end;
pcm_channels = bebob->tx_stream_formations[index].pcm;
midi_channels = bebob->tx_stream_formations[index].midi;
err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
pcm_channels, midi_channels * 8,
false);
if (err < 0)
goto end;
pcm_channels = bebob->rx_stream_formations[index].pcm;
midi_channels = bebob->rx_stream_formations[index].midi;
err = amdtp_am824_set_parameters(&bebob->rx_stream, rate,
pcm_channels, midi_channels * 8,
false);
if (err < 0)
goto end;
/* 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)
{
cmp_connection_destroy(&bebob->in_conn);
cmp_connection_destroy(&bebob->out_conn);
}
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_am824_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;
}
/*
* BeBoB v3 transfers packets with these qurks:
* - In the beginning of streaming, the value of dbc is incremented
* even if no data blocks are transferred.
* - The value of dbc is reset suddenly.
*/
if (bebob->version > 2)
bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC |
CIP_SKIP_DBC_ZERO_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_am824_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);
}
end:
return err;
}
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
{
const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
unsigned int curr_rate;
int err = 0;
/* Need no substreams */
if (bebob->substreams_counter == 0)
goto end;
/*
* Considering JACK/FFADO streaming:
* TODO: This can be removed hwdep functionality becomes popular.
*/
err = check_connection_used_by_others(bebob, &bebob->rx_stream);
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(&bebob->rx_stream))
amdtp_stream_stop(&bebob->rx_stream);
if (amdtp_streaming_error(&bebob->tx_stream))
amdtp_stream_stop(&bebob->tx_stream);
if (!amdtp_stream_running(&bebob->rx_stream) &&
!amdtp_stream_running(&bebob->tx_stream))
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(&bebob->rx_stream);
amdtp_stream_stop(&bebob->tx_stream);
break_both_connections(bebob);
}
/* master should be always running */
if (!amdtp_stream_running(&bebob->rx_stream)) {
/*
* 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, &bebob->rx_stream, 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(&bebob->rx_stream);
break_both_connections(bebob);
goto end;
}
}
/* wait first callback */
if (!amdtp_stream_wait_callback(&bebob->rx_stream,
CALLBACK_TIMEOUT)) {
amdtp_stream_stop(&bebob->rx_stream);
break_both_connections(bebob);
err = -ETIMEDOUT;
goto end;
}
}
/* start slave if needed */
if (!amdtp_stream_running(&bebob->tx_stream)) {
err = start_stream(bebob, &bebob->tx_stream, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to run AMDTP slave stream:%d\n", err);
amdtp_stream_stop(&bebob->rx_stream);
break_both_connections(bebob);
goto end;
}
/* wait first callback */
if (!amdtp_stream_wait_callback(&bebob->tx_stream,
CALLBACK_TIMEOUT)) {
amdtp_stream_stop(&bebob->tx_stream);
amdtp_stream_stop(&bebob->rx_stream);
break_both_connections(bebob);
err = -ETIMEDOUT;
}
}
end:
return err;
}
void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
{
if (bebob->substreams_counter == 0) {
amdtp_stream_pcm_abort(&bebob->rx_stream);
amdtp_stream_stop(&bebob->rx_stream);
amdtp_stream_pcm_abort(&bebob->tx_stream);
amdtp_stream_stop(&bebob->tx_stream);
break_both_connections(bebob);
}
}
/*
* This function should be called before starting streams or after stopping
* streams.
*/
void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
{
amdtp_stream_destroy(&bebob->rx_stream);
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
}
/*
* 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 == ARRAY_SIZE(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 > AM824_MAX_CHANNELS_FOR_PCM ||
formation[i].midi > AM824_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)
{
const 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);
}