OpenCloudOS-Kernel/sound/firewire/bebob/bebob_stream.c

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// SPDX-License-Identifier: GPL-2.0-only
/*
* bebob_stream.c - a part of driver for BeBoB based devices
*
* Copyright (c) 2013-2014 Takashi Sakamoto
*/
#include "./bebob.h"
#define CALLBACK_TIMEOUT 2500
ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits packets with discontinuous value in dbc field. This causes two situation, one is to abort streaming by firewire-lib as a result of detecting the discontinuity. Another is to call driver's .update() because of bus reset. These two is generated independently. (The former depends on isochronous stream and the latter depends on IEEE1394 bus driver.) When BeBoB driver works with XRUN-recoverable applications, this situation looks like stream_start_duplex() call followed by stream_update_duplex() call because applications will call snd_pcm_prepare() immediately at XRUN. To update connections and streams at first, this commit use completion. When queueing error occurs, stream_start_duplex() is forced to wait maximum 1000msec. During this, when .update() is called, the completion is waken and stream_start_duplex() is processed without breaking connections. At bus reset, stream_start_duplex() shouldn't break/establish connections and stream_update_duplex() should update connections because a caller of fw_iso_resources_allocate() is responsible for calling fw_iso_resources_update() on bus reset. This commit also adds a flag, which has an effect to skip checking continuity for first packet. This flag is useful for BeBoB quirk to start handling packets during streaming. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-25 21:45:16 +08:00
#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;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
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;
}
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
/* 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;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
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;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
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;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
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;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_SYNC) {
/* Often word clock. */
*src = SND_BEBOB_CLOCK_TYPE_EXTERNAL;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
goto end;
} else if (type == AVC_BRIDGECO_PLUG_TYPE_ADDITION) {
/*
* Not standard.
* Mostly, additional internal clock.
*/
*src = SND_BEBOB_CLOCK_TYPE_INTERNAL;
ALSA: bebob: improve signal mode detection for clock source With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-14 11:49:27 +08:00
goto end;
}
}
}
/* Not supported. */
err = -EIO;
end:
return err;
}
static 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;
ALSA: bebob: Add support for M-Audio usual Firewire series This commit allows this driver to support some models which M-Audio produces with DM1000/DM1000E with usual firmware. They are: - Firewire 410 - Firewire AudioPhile - Firewire Solo - Ozonic - NRV10 - FirewireLightBridge According to a person who worked in BridgeCo, some models are produced with 'Pre-BeBoB'. This means that these products were released before BeBoB was officially produced, and later BeBoB specification was formed. So these models have some quirks. M-Audio usual firmware quirks: - Just after powering on, 'Firewire 410' waits to download firmware. This state is changed when receiving cue. Then bus reset is generated and the device is recognized as a different model with the uploaded firmware. - 'Firewire Audiophile' also waits to download firmware but its vendor id/model id is the same as the one after loading firmware. - The information of channel mapping for MIDI conformant data channel is invalid against BridgeCo specification. This commit adds some codes for these quirks but don't support to upload firmware. This commit also adds specific operations to get metering information. The metering information also includes status of clock synchronization if the model supports to switch source of clock. The specification of FirewireLightBridge is unknown. So in this time, normal operations are applied for this model. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-25 21:45:25 +08:00
/*
* 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
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 void break_both_connections(struct snd_bebob *bebob)
{
cmp_connection_break(&bebob->in_conn);
cmp_connection_break(&bebob->out_conn);
ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits packets with discontinuous value in dbc field. This causes two situation, one is to abort streaming by firewire-lib as a result of detecting the discontinuity. Another is to call driver's .update() because of bus reset. These two is generated independently. (The former depends on isochronous stream and the latter depends on IEEE1394 bus driver.) When BeBoB driver works with XRUN-recoverable applications, this situation looks like stream_start_duplex() call followed by stream_update_duplex() call because applications will call snd_pcm_prepare() immediately at XRUN. To update connections and streams at first, this commit use completion. When queueing error occurs, stream_start_duplex() is forced to wait maximum 1000msec. During this, when .update() is called, the completion is waken and stream_start_duplex() is processed without breaking connections. At bus reset, stream_start_duplex() shouldn't break/establish connections and stream_update_duplex() should update connections because a caller of fw_iso_resources_allocate() is responsible for calling fw_iso_resources_update() on bus reset. This commit also adds a flag, which has an effect to skip checking continuity for first packet. This flag is useful for BeBoB quirk to start handling packets during streaming. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-25 21:45:16 +08:00
// These models seem to be in transition state for a longer time. When
// accessing in the state, any transactions is corrupted. In the worst
// case, the device is going to reboot.
if (bebob->version < 2)
msleep(600);
}
static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream)
{
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)
return err;
}
err = cmp_connection_establish(conn);
if (err < 0)
return err;
return amdtp_domain_add_stream(&bebob->domain, stream,
conn->resources.channel, conn->speed);
}
static int init_stream(struct snd_bebob *bebob, struct amdtp_stream *stream)
{
enum amdtp_stream_direction dir_stream;
struct cmp_connection *conn;
enum cmp_direction dir_conn;
int err;
if (stream == &bebob->tx_stream) {
dir_stream = AMDTP_IN_STREAM;
conn = &bebob->out_conn;
dir_conn = CMP_OUTPUT;
} else {
dir_stream = AMDTP_OUT_STREAM;
conn = &bebob->in_conn;
dir_conn = CMP_INPUT;
}
err = cmp_connection_init(conn, bebob->unit, dir_conn, 0);
if (err < 0)
return err;
err = amdtp_am824_init(stream, bebob->unit, dir_stream, CIP_BLOCKING);
if (err < 0) {
cmp_connection_destroy(conn);
return err;
}
ALSA: bebob: simplify bus-reset handling At bus-reset, DM1000/DM1100/DM1500 chipsets transfer packets with discontinuous value in 'dbc' field of CIP header. In this case, packet streaming layer in firewire-lib module stops streaming and set XRUN to PCM substream. In ALSA, PCM applications are notified the XRUN status by the return value of ALSA PCM interface. They can recover this state by executing snd_pcm_prepare(), then PCM drivers' prepare handler is called, and start new PCM substream. For ALSA BeBoB driver, the handler establishes new connections and start new AMDTP streaming. Unfortunately, neither the PCM applications nor the driver know the reason of XRUN. The driver gets to know the reason when update handler is called by IEEE 1394 bus driver. As long as I tested, the order of below events are not fixed: * Detecting packet discontinuity in tasklet context of OHCI 1394 driver * Calling prepare handler in process context of ALSA PCM application * Calling update handler in kthread context of IEEE 1394 bus driver The unpredictable order is disadvantage for the driver to be compliant to CMP. In IEC 61883-1, new CMP establish operations should be done 1 sec (isoc_resource_delay) after bus-reset. Within 1 sec, CMP restore operations are allowed. For this reason, in former commit ('b6bc812327aa: ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset'), the process context is forced to wait for executing update handler. The process context wait for bus-reset up to 1 sec. This commit solves the issue, while causes more disadvantages. For PCM applications, calling snd_pcm_prepare() for recovering XRUN state takes more time and the driver got a bit complicated code, while the recovery is not always successful. As long as I tested, DM1000/DM1100/DM1500 and BeBoB firmware can allow drivers to establish new connections just after bus reset. Furthermore, any FCP transactions are handled correctly. Therefore, the driver don't need to wait for bus reset handler for starting new streaming. This commit removes the codes to reduce maintenance cost. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-02-20 15:18:56 +08:00
if (stream == &bebob->tx_stream) {
// 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;
}
return 0;
}
static void destroy_stream(struct snd_bebob *bebob, struct amdtp_stream *stream)
{
amdtp_stream_destroy(stream);
if (stream == &bebob->tx_stream)
cmp_connection_destroy(&bebob->out_conn);
else
cmp_connection_destroy(&bebob->in_conn);
}
int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
{
int err;
err = init_stream(bebob, &bebob->tx_stream);
if (err < 0)
return err;
err = init_stream(bebob, &bebob->rx_stream);
if (err < 0) {
destroy_stream(bebob, &bebob->tx_stream);
return err;
}
err = amdtp_domain_init(&bebob->domain);
if (err < 0) {
destroy_stream(bebob, &bebob->tx_stream);
destroy_stream(bebob, &bebob->rx_stream);
}
return err;
}
static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream,
unsigned int rate, unsigned int index)
{
struct snd_bebob_stream_formation *formation;
struct cmp_connection *conn;
int err;
if (stream == &bebob->tx_stream) {
formation = bebob->tx_stream_formations + index;
conn = &bebob->out_conn;
} else {
formation = bebob->rx_stream_formations + index;
conn = &bebob->in_conn;
}
err = amdtp_am824_set_parameters(stream, rate, formation->pcm,
formation->midi, false);
if (err < 0)
return err;
return cmp_connection_reserve(conn, amdtp_stream_get_max_payload(stream));
}
int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate,
unsigned int frames_per_period,
unsigned int frames_per_buffer)
{
unsigned int curr_rate;
int err;
// 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)
return err;
err = bebob->spec->rate->get(bebob, &curr_rate);
if (err < 0)
return err;
if (rate == 0)
rate = curr_rate;
if (curr_rate != rate) {
amdtp_domain_stop(&bebob->domain);
ALSA: bebob/firewire-lib: Add a quirk for discontinuity at bus reset Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits packets with discontinuous value in dbc field. This causes two situation, one is to abort streaming by firewire-lib as a result of detecting the discontinuity. Another is to call driver's .update() because of bus reset. These two is generated independently. (The former depends on isochronous stream and the latter depends on IEEE1394 bus driver.) When BeBoB driver works with XRUN-recoverable applications, this situation looks like stream_start_duplex() call followed by stream_update_duplex() call because applications will call snd_pcm_prepare() immediately at XRUN. To update connections and streams at first, this commit use completion. When queueing error occurs, stream_start_duplex() is forced to wait maximum 1000msec. During this, when .update() is called, the completion is waken and stream_start_duplex() is processed without breaking connections. At bus reset, stream_start_duplex() shouldn't break/establish connections and stream_update_duplex() should update connections because a caller of fw_iso_resources_allocate() is responsible for calling fw_iso_resources_update() on bus reset. This commit also adds a flag, which has an effect to skip checking continuity for first packet. This flag is useful for BeBoB quirk to start handling packets during streaming. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-25 21:45:16 +08:00
break_both_connections(bebob);
cmp_connection_release(&bebob->out_conn);
cmp_connection_release(&bebob->in_conn);
}
if (bebob->substreams_counter == 0 || curr_rate != rate) {
unsigned int index;
// 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.
err = bebob->spec->rate->set(bebob, rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to set sampling rate: %d\n",
err);
return err;
}
err = get_formation_index(rate, &index);
if (err < 0)
return err;
err = keep_resources(bebob, &bebob->tx_stream, rate, index);
if (err < 0)
return err;
err = keep_resources(bebob, &bebob->rx_stream, rate, index);
if (err < 0) {
cmp_connection_release(&bebob->out_conn);
return err;
}
err = amdtp_domain_set_events_per_period(&bebob->domain,
frames_per_period, frames_per_buffer);
if (err < 0) {
cmp_connection_release(&bebob->out_conn);
cmp_connection_release(&bebob->in_conn);
return err;
}
}
return 0;
}
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob)
{
int err;
// Need no substreams.
if (bebob->substreams_counter == 0)
return -EIO;
// packet queueing error or detecting discontinuity
if (amdtp_streaming_error(&bebob->rx_stream) ||
amdtp_streaming_error(&bebob->tx_stream)) {
amdtp_domain_stop(&bebob->domain);
break_both_connections(bebob);
}
if (!amdtp_stream_running(&bebob->rx_stream)) {
enum snd_bebob_clock_type src;
struct amdtp_stream *master, *slave;
unsigned int curr_rate;
unsigned int ir_delay_cycle;
if (bebob->maudio_special_quirk) {
err = bebob->spec->rate->get(bebob, &curr_rate);
if (err < 0)
return err;
}
err = snd_bebob_stream_get_clock_src(bebob, &src);
if (err < 0)
return err;
if (src != SND_BEBOB_CLOCK_TYPE_SYT) {
master = &bebob->tx_stream;
slave = &bebob->rx_stream;
} else {
master = &bebob->rx_stream;
slave = &bebob->tx_stream;
}
err = start_stream(bebob, master);
if (err < 0)
goto error;
err = start_stream(bebob, slave);
if (err < 0)
goto error;
// The device postpones start of transmission mostly for 1 sec
// after receives packets firstly. For safe, IR context starts
// 0.4 sec (=3200 cycles) later to version 1 or 2 firmware,
// 2.0 sec (=16000 cycles) for version 3 firmware. This is
// within 2.5 sec (=CALLBACK_TIMEOUT).
// Furthermore, some devices transfer isoc packets with
// discontinuous counter in the beginning of packet streaming.
// The delay has an effect to avoid detection of this
// discontinuity.
if (bebob->version < 2)
ir_delay_cycle = 3200;
else
ir_delay_cycle = 16000;
err = amdtp_domain_start(&bebob->domain, ir_delay_cycle);
if (err < 0)
goto error;
// NOTE:
// The firmware customized by M-Audio uses these commands to
// start transmitting stream. This is not usual way.
if (bebob->maudio_special_quirk) {
err = bebob->spec->rate->set(bebob, curr_rate);
if (err < 0) {
dev_err(&bebob->unit->device,
"fail to ensure sampling rate: %d\n",
err);
goto error;
}
}
if (!amdtp_stream_wait_callback(&bebob->rx_stream,
CALLBACK_TIMEOUT) ||
!amdtp_stream_wait_callback(&bebob->tx_stream,
CALLBACK_TIMEOUT)) {
err = -ETIMEDOUT;
goto error;
}
}
return 0;
error:
amdtp_domain_stop(&bebob->domain);
break_both_connections(bebob);
return err;
}
void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
{
if (bebob->substreams_counter == 0) {
amdtp_domain_stop(&bebob->domain);
break_both_connections(bebob);
cmp_connection_release(&bebob->out_conn);
cmp_connection_release(&bebob->in_conn);
}
}
/*
* This function should be called before starting streams or after stopping
* streams.
*/
void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob)
{
amdtp_domain_destroy(&bebob->domain);
destroy_stream(bebob, &bebob->tx_stream);
destroy_stream(bebob, &bebob->rx_stream);
}
/*
* 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);
}