On Fireworks board module of Echo Audio, TSB43Cx43A (IceLynx Micro, iCEM)
is used to process payload of isochronous packets. There's an public
document of this chip[1]. This document is for firmware programmers to
transfer/receive AMDTP with IEC60958 data format, however in clause 2.5,
2.6 and 2.7, we can see system design to utilize the sequence of value in
SYT field of CIP header. In clause 2.3, we can see the specification of
Audio Master Clock (MCLK) from iCEM.
Well, this clock is actually not used for sampling clock. This can be
confirmed when corresponding driver transfer random value as the sequence
of SYT field. Even if in this case, the unit generates proper sound.
Additionally, in unique command set for this board module, the format
of CIP is changed; for IEC 61883-6 mode which we use, and for Windows
Operating System. In the latter mode, the whole 32 bit field in second CIP
header from Windows driver is used to represent counter of packets (NO-DATA
code is still used for packets without data blocks). If the master clock
was physically used by DSP on the board module, the Windows driver must
have transferred correct sequence of SYT field.
Furthermore, as long as seeing capacities of AudioFire2, AudioFire4,
AudioFire8, AudioFirePre8 and AudioFire12, these models don't support
SYT-Match clock source.
Summary, we have no need to relate incoming/outgoing packets. This commit
drops reusing SYT sequence of incoming packets for outgoing packets.
[1] Using TSB43Cx43A: S/PDIF over 1394 (2003, Texus Instruments
Incorporated)
http://www.ti.com/analog/docs/litabsmultiplefilelist.tsp?literatureNumber=slla148&docCategoryId=1&familyId=361
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows driver for BeBoB-based models mostly wait for transmitted packets,
then transfer packets to the models. This looks for the relationship
between incoming packets and outgoing packets to synchronize the sequence
of presentation timestamp.
However, the sequence between packets of both direction has no
relationship. Even if receiving NO-DATA packets, the drivers transfer
packets with meaningful value in SYT field. Additionally, the order of
starting packets is always the same, independently of the source of clock.
The corresponding driver is expected as a generator of presentation
timestamp and these models can select it as a source of sampling clock.
This commit drops reusing SYT sequence from ALSA bebob driver. The driver
always transfer packets with presentation timestamp generated by ALSA
firewire stack, without re-using the sequence of value in SYT field in
incoming packets to outgoing packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When audio and music units have some quirks in their sequence of packet,
it's really hard for non-owners to identify the quirks. Although developers
need dumps for sequence of packets, it's difficult for users who have no
knowledges and no equipments for this purpose.
This commit adds tracepoints for this situation. When users encounter
the issue, they can dump a part of packet data via Linux tracing framework
as long as using drivers in ALSA firewire stack.
Additionally, tracepoints for outgoing packets will be our help to check
and debug packet processing of ALSA firewire stack.
This commit newly adds 'snd_firewire_lib' subsystem with 'in_packet' and
'out_packet' events. In the events, some attributes of packets and the
index of packet managed by this module are recorded per packet.
This is an usage:
$ trace-cmd record -e snd_firewire_lib:out_packet \
-e snd_firewire_lib:in_packet
/sys/kernel/tracing/events/snd_firewire_lib/out_packet/filter
/sys/kernel/tracing/events/snd_firewire_lib/in_packet/filter
Hit Ctrl^C to stop recording
^C
$ trace-cmd report trace.dat
...
23647.033934: in_packet: 01 4073 ffc0 ffc1 00 000f0040 9001b2d1 122 44
23647.033936: in_packet: 01 4074 ffc0 ffc1 00 000f0048 9001c83b 122 45
23647.033937: in_packet: 01 4075 ffc0 ffc1 00 000f0050 9001ffff 002 46
23647.033938: in_packet: 01 4076 ffc0 ffc1 00 000f0050 9001e1a6 122 47
23647.035426: out_packet: 01 4123 ffc1 ffc0 01 010f00d0 9001fb40 122 17
23647.035428: out_packet: 01 4124 ffc1 ffc0 01 010f00d8 9001ffff 002 18
23647.035429: out_packet: 01 4125 ffc1 ffc0 01 010f00d8 900114aa 122 19
23647.035430: out_packet: 01 4126 ffc1 ffc0 01 010f00e0 90012a15 122 20
(Here, some common fields are omitted so that a line to be within 80
characters.)
...
One line represent one packet. The legend for the last nine fields is:
- The second of cycle scheduled for the packet
- The count of cycle scheduled for the packet
- The ID of node as source (hex)
- Some devices transfer packets with invalid source node ID in their CIP
header.
- The ID of node as destination (hex)
- The value is not in CIP header of packets.
- The value of isochronous channel
- The first quadlet of CIP header (hex)
- The second quadlet of CIP header (hex)
- The number of included quadlets
- The index of packet in a buffer maintained by this module
This is an example to parse these lines from text file by Python3 script:
\#!/usr/bin/env python3
import sys
def parse_ts(second, cycle, syt):
offset = syt & 0xfff
syt >>= 12
if cycle & 0x0f > syt:
cycle += 0x10
cycle &= 0x1ff0
cycle |= syt
second += cycle // 8000
cycle %= 8000
# In CYCLE_TIMER of 1394 OHCI, second is represented in 8 bit.
second %= 128
return (second, cycle, offset)
def calc_ts(second, cycle, offset):
ts = offset
ts += cycle * 3072
# In DMA descriptor of 1394 OHCI, second is represented in 3 bit.
ts += (second % 8) * 8000 * 3072
return ts
def subtract_ts(minuend, subtrahend):
# In DMA descriptor of 1394 OHCI, second is represented in 3 bit.
if minuend < subtrahend:
minuend += 8 * 8000 * 3072
return minuend - subtrahend
if len(sys.argv) != 2:
print('At least, one argument is required for packet dump.')
sys.exit()
filename = sys.argv[1]
data = []
prev = 0
with open(filename, 'r') as f:
for line in f:
pos = line.find('packet:')
if pos < 0:
continue
pos += len('packet:')
line = line[pos:].strip()
fields = line.split(' ')
datum = []
datum.append(fields[8])
syt = int(fields[6][4:], 16)
# Empty packet in IEC 61883-1, or NODATA in IEC 61883-6
if syt == 0xffff:
data_blocks = 0
else:
payload_size = int(fields[7], 10)
data_block_size = int(fields[5][2:4], 16)
data_blocks = (payload_size - 2) / data_block_size
datum.append(data_blocks)
second = int(fields[0], 10)
cycle = int(fields[1], 10)
start = (second << 25) | (cycle << 12)
datum.append('0x{0:08x}'.format(start))
start = calc_ts(second, cycle, 0)
datum.append("0x" + fields[5])
datum.append("0x" + fields[6])
if syt == 0xffff:
second = 0
cycle = 0
tick = 0
else:
second, cycle, tick = parse_ts(second, cycle, syt)
ts = calc_ts(second, cycle, tick)
datum.append(start)
datum.append(ts)
if ts == 0:
datum.append(0)
datum.append(0)
else:
# Usual case, or a case over 8 seconds.
if ts > start or start > 7 * 8000 * 3072:
datum.append(subtract_ts(ts, start))
if ts > prev or start > 7 * 8000 * 3072:
gap = subtract_ts(ts, prev)
datum.append(gap)
else:
datum.append('backward')
else:
datum.append('invalid')
prev = ts
data.append(datum)
sys.exit()
The data variable includes array with these elements:
- The index of the packet
- The number of data blocks in the packet
- The value of cycle count (hex)
- The value of CIP header 1 (hex)
- The value of CIP header 2 (hex)
- The value of cycle count (tick)
- The value of calculated presentation timestamp (tick)
- The offset between the cycle count and presentation timestamp
- The elapsed ticks from the previous presentation timestamp
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In callback function of isochronous context, modules can queue packets to
indicated isochronous cycles. Although the cycle to queue a packet is
deterministic by calculation, this module doesn't implement the calculation
because it's useless for processing.
In future, the cycle count is going to be printed with the other parameters
for debugging. This commit is the preparation. The cycle count is computed
by cycle unit, and correctly arranged to corresponding packets. The
calculated count is used in later commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In callback function of isochronous context, u32 variable is passed for
cycle count. The value of this variable comes from DMA descriptors of 1394
Open Host Controller Interface (1394 OHCI). In the specification, DMA
descriptors transport lower 3 bits for second field and full cycle field in
16 bits field, therefore 16 bits of the u32 variable are available. The
value for second is modulo 8, and the value for cycle is modulo 8,000.
Currently, ALSA firewire-lib module don't use the value of the second
field, because the value is useless to calculate presentation timestamp in
IEC 61883-6. However, the value may be useful for debugging. In later
commit, it will be printed with the other parameters for debugging.
This commit makes this module to handle the whole cycle count including
second. The value is calculated by cycle unit. The existed code is already
written with ignoring the value of second, thus this commit causes no
issues.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When some tascam units are connected sequentially, userspace
applications are involved at bus-reset state on IEEE 1394 bus. In the
state, any communications can be canceled. Therefore, sound card
registration should be delayed till the bus gets calm.
This commit achieves it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When some digi00x units are connected sequentially, userspace
applications are involved at bus-reset state on IEEE 1394 bus. In the
state, any communications can be canceled. Therefore, sound card
registration should be delayed till the bus gets calm.
This commit achieves it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some oxfw based units tends to fail asynchronous communication when
IEEE 1394 bus is under bus-reset state. When registering sound card
instance at unit probe callback, userspace applications can be involved
to the state.
This commit postpones the registration till the bus is calm.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When some fireworks units are connected sequentially, userspace
applications are involved at bus-reset state on IEEE 1394 bus. In the
state, any communications can be canceled. Therefore, sound card
registration should be delayed till the bus gets calm.
This commit achieves it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some bebob based units tends to fail asynchronous communication when
IEEE 1394 bus is under bus-reset state. When registering sound card
instance at unit probe callback, userspace applications can be involved
to the state.
This commit postpones the registration till the bus is calm.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, ALSA dice driver postpone sound card registration after
IEEE 1394 bus is calm. This idea has advantages for the other drivers.
This commit adds a helper function for it to firewire-lib module. The
function is really for the specific purpose. Callers should initialize
delayed work structure with callback function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, ALSA dice driver doesn't generate kernel warnings
when unplugging units before initializing stream data.
This commit moves the initialization to delayed registration of sound
card, to simplify unit probe processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When any of AMDTP stream data are not initialized and private data is
going to be released, WARN_ON() in amdtp_stream_destroy() is hit and
dump messages. This may take users irritated.
This commit fixes the bug to skip releasing when it's not initialized.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot it.
Fixes: 3e78e1518e12('ALSA: firewire-tascam: add support for FW-1804')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, model-specific quirks are detected out of information parser,
however it's natural to detect it in the parser.
This commit applies the idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, 'struct device_info' is obsoleted, whereas private
structure still keeps a pointer to it.
This commit remove the member.
d6ce6bbd7d83('ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'vendor_id' argument is not used in the local function. Let's remove
it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When sound card is going to be released, dice private data is
also released. Then all of data should be released. However,
stream data is not released. This causes memory leak when
unplugging dice unit.
This commit fixes the bug.
Fixes: 4bdc495c87b3('ALSA: dice: handle several PCM substreams when any isochronous streams are available')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In dice interface, two blocks of register are accessible via IEEE 1394
asynchronous transaction to represent the number of supported isochronous
streams and the number of quadlets for stream information.
Current ALSA dice driver uses array with 'unsigned int' element for
temporary cache of these information. But using structure is preferable
for begin easily comprehensible.
This commit applies a local structure for this aim.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models reduce the number of available isochronous streams for higher
sampling transfer frequency. Such models bring an issue about how to add
PCM substreams. When at lower sampling transfer frequency, the
models reports whole available streams, thus this driver can add enough
number of PCM substreams at probing time. On the other hand, at higher
sampling transfer frequency, this driver can just add reduced number of
PCM substreams. After probed, even if the sampling transfer frequency is
changed to lower rate, fewer PCM substreams are actually available. This
is inconvenience.
For the reason, this commit adds a list so that this driver assume models
on the list to have two pairs of PCM substreams. This list keeps the name
of model in which the number of available streams differs depending on
sampling transfer frequency.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, ALSA dice driver can handle available isochronous
streams. This commit adds support for several PCM substreams on the
streams.
The additional PCM substreams are available via another ALSA PCM character
devices so that one ALSA PCM application can handle them without cumbersome
operations. For example, two PCM substreams are available on each stream,
two ALSA character devices are added for them. In configuration space of
alsa-lib, it's represented with 'hw:0,0' and 'hw:0,1'.
The PCM substreams are constraint to parameters of the corresponding
streams. If the PCM substreams are unavailable for some reasons,
open(2) to ALSA PCM character device returns error and reports ENXIO.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit enables ALSA dice driver to handle whole available streams.
In Dice, certain registers represent the number of available streams at
current sampling transfer frequency for both directions. The parameters
of each stream are represented in a block of register. This block is
aligned sequentially. These streams start simultaneously by writing
enable bit to a register.
This commit operates these registers when starting/stopping streams.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently ALSA dice driver handles a pair of isochronous resources for
IEC 61883-1/6 packet streaming. While, according to some documents about
ASICs named as 'Dice', several isochronous streams are available.
Here, I start to describe ASICs produced under 'Dice' name.
* Dice II (designed by wavefront semiconductor, including TCAT's IP)
* STD (with limited functionality of DTCP)
* CP (with full functionality of DTCP)
* TCD2210/2210-E (so-called 'Dice Mini')
* TCD2220/2220-E (so-called 'Dice Jr.')
* TCD3070-CH (so-called 'Dice III')
Some documents are public and we can see hardware design of them. We can
find some articles about hardware internal register definitions
(not registers exported to IEEE 1394 bus).
* DICE II User Guide
* http://www.tctechnologies.tc/archive/downloads/dice_ii_user_guide.pdf
* 6.1 AVS Audio Receivers
* Table 6.1: AVS Audio Receiver Memory Map
* ARX1-ARX4
* 6.2 AVS Audio Transmitters
* Table 6.2: AVS Audio Transmitter Memory Map
* ATX1, ATX2
* TCD22xx User Guide
* http://www.tctechnologies.tc/downloads/tcd22xx_user_guide.pdf
* 6.1 AVS Audio Receivers
* Table 66: AVS Audio Receiver Memory Map
* ARX1, ARX2
* 6/2 AVS Audio Transmitters
* Table 67: AVS Audio Transmitter Memory Map
* ATX1, ATX2
* DICE III
* http://www.tctechnologies.tc/downloads/TCD3070-CH.pdf
* Dual stream 63 channel transmitter/receiver
For Dice II and TCD22xx series, maximum 16 data channels are transferred in
an AMDTP packet, while for Dice III, maximum 32 data channels are
transferred.
According to the design of the series of these ASICs, this commit allows
this driver to handle additional set of isochronous resources. For
practical reason, two pair of isochronous resources are added. As of this
commit, this driver still use a pair of the first isochronous resources.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit drops implementation of duplex streams synchronization
from ALSA dice driver, due to a reason of hardware design. This patch
allows dice-based units to generate sounds correctly when isochronous
packet streaming starts at first time.
In IEC 61883-6:2005, CIP packetization layer for AM824 data format
utilizes the value of SYT field in CIP header of received packet for
a reference to phase lock loop. Figure 3 in clause 4.3 describes it.
The value is an offset from cycle_time field of every cycle start packet
from cycle master on IEEE 1394 bus. The time calculated with these two
fields is called as 'presentation timestamp' which represents the time
to play data included in the packet.
Although, this idea includes some problems due to accuracy of timekeep in
cycle master, accuracy of transmission of cycle start packet on the bus
with the other units, accuracy of sampling clock in data transmitter side
and accuracy of replay in data receiver side. In most case, these
accuracies somewhat worse because there's no such ideal hardwares in this
world.
For the issues, ASICs for Dice include Jitter Elimination Technologies
(JET) PLL. The PLL can handle several sources of clock and compensate it
with high-precision internal clock source. The sequence of value in syt
field of received AMDTP packets is one of the sources, therefore
transmitters on IEEE 1394 bus should transfer it.
On the other hand, current ALSA dice driver is programmed with a mode of
duplex streams with synchronization. In this mode, the driver outputs
packets after some incoming packets are handled, to re-use the value of
SYT field in incoming packets to the value for outgoing packets. This mode
is enabled when source signal of sampling clock is set to internal, and
this is a major use case. Thus, in most cases, the unit receives no packets
during a short time after packet streaming starts.
As long as I experienced, this causes the units to generate no sounds at
first time to receive packets. This issue occurs only with Dice II. I guess
this is due to a quirk of the PLL. In short, the PLL cannot generate firm
signals to ADCs/DACs or the other ICs when no packets are received in the
beginning of packet streaming. While, on second time or later, the unit
generates sound correctly. I guess that starting packet streaming at first
time sets the PLL correctly.
Well, still based on my hypothesis and no way to prove it, this commit
drops duplex streams synchronization from this driver. At least, the PLL
requires the sequence of value in SYT field of received AMDTP packets as
one of source of clock signals with internal clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a previous commit, ALSA oxfw driver retries transferring MIDI
messages at transaction failure for scs1x. On the other hand, there're
fatal transaction error. Then, no MIDI messages reach to the unit anymore.
In this case, MIDI substream should be terminated.
This commit stops MIDI transmission after the fatal error occurs.
Unfortunately, unlike ALSA PCM functionality, ALSA rawmidi core has no
feature to discontinue MIDI substream runtime in kernel side, thus this
commit just stops MIDI transmission without notifying it to userspace.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, ALSA oxfw driver has a TODO to retry MIDI transferring
at transaction failure.
This commit achieves it. Current implementation uses snd_rawmidi_transmit()
to transfer messages, thus the target MIDI messages are not in buffer when
transaction failure is detected. Although we cannot use a pair of
snd_rawmidi_transmit_peek() and snd_ramwidi_transmit_ack(), the
messages are still in scs1x specific structure and the data is available
for retries.
This commit adds a member to the structure for the length of buffered
messages, and uses the value again at retries.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1, at bus-reset, applications can continue isochronous
streaming by updating connections. In ALSA fireworks driver, the
operation is executed in 'update' handler for bus driver.
The connection resources are also changed in process contexts of PCM/MIDI
applications. Therefore, bus-reset handling has race condition
against connection. Current ALSA fireworks driver has a bug for the
condition.
This commit fixes the bug, by expand critical section with mutex. As a
result, connection updating operation in bus-reset handler and connection
changing operation in process context are serialized.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DM1000/DM1100/DM1500 chipsets transfer packets with discontinue value in
'dbc' field of CIP header. For ALSA bebob driver, this makes its bus-reset
handler meaningless, because the discontinuity is detected quite earlier
than executing the handler.
This commit gives up updating streams at the bus reset handler.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The counter is incremented/decremented in critical section protected with
mutex. Therefore, no need to use atomic_t.
This commit changes the type to unsigned int.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, critical section is protected by mutex in functions of
fireworks_stream.c. Callers increments/decrements substreams counter
before calling the functions. Moving mutex to the callers code allows
to change type of the substream counter from atomic_t to unsigned int.
This commit is a preparation for obsoleting usage of atomic_t for
substream counter.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This commit replaces tasklet with workqueue for scs1x functionality of
ALSA oxfw driver.
This driver transfers MIDI message specific for SCS.1m and SCS.1d. This
task is currently done in software IRQ context of tasklet. In a view of
system, this context is limited resources and some important drivers (at
least, more important than ALSA oxfw driver) use the context as its
bottom-harf.
If the work to transfer MIDI messages is done within a time, it's better
to use the other context for the work. Actually, with recent CPUs, the
work will be scheduled within a time. This is a reason of this commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I tested, Dice-based models produced by TC Electronic with
factory-configured settings transfer no notification within
ensure_phase_lock(). On the other hand, with upgraded firmwares, it
starts to transfer the notification. This seems to be a quirk of earlier
firmwares.
This commit ensures phase lock by reading a register after waiting for
the notification. Even if it's timed-out, ensure_phase_lock() return
success as long as the register has expected clock status.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With former patchset, ALSA dice driver doesn't change clock parameters
anymore, while the driver still touch clock configuration for phase lock.
Although the locking status is in Dice notification, the driver doesn't
detect it. Usually, this causes no issues because in most case
NOTIFY_LOCK_CHG notification transfers after NOTIFY_CLOCK_ACCEPTED
notification, while it's better to detect locking status.
This commit changes notification mask just to detect lock status change.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, probing process has no need to set sampling transfer
frequency. Although it's OK to drop a function to change the frequency
from this module, some models require it before streaming. This seems to
be due to phase lock of clock source.
This commit moves the function from transaction layer to stream layer, and
rename it according to the purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dice interface design doesn't allow drivers to read supported combination
between sampling transfer frequencies and the number of Multi bit linear
audio data channels. Due to the design, ALSA dice driver changes current
sampling transfer frequency to generate cache of the combinations at
device probing processing.
Although, this idea is worse because ALSA dice driver changes the state of
clock. This is not what users want when they save favorite configuration
to the device in advance.
Furthermore, there's a possibility that the format of data block is decided
not only according to current sampling transfer frequency, but also the
other factors, i.e. data format for digital interface. It's not good to
generate channel cache according to the sampling transfer frequency only.
This commit purges processing cache data and related structure members. As
a result, users must set preferable sampling transfer frequency before
using ALSA PCM applications, as long as they want to start any PCM
substreams at the rate except for current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to remove members related to channel cache
for the number of channels for multi bit linear audio data and MIDI
ports. This commit changes the way to get the number of multi bit linear
audio data channel. It's directly retrieved by asynchronous transactions
to some registers.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes the way to add ALSA MIDI ports. This driver read the
number of multiplexed MIDI substreams from hardware register, then adds the
same number of ALSA MIDI ports. This commit is based on my assumption that
the number is fixed at all of supported sampling transfer frequency.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, ALSA Dice driver limits PCM substreams at current
sampling transfer frequency and current number of Multi bit linear audio
data channel. Thus, the driver has no need to start AMDTP streams at
the other sampling transfer frequency except for current one. This is due
to Dice interface design.
This commit limits AMDTP stream at current sampling transfer frequency,
according to the design.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM core has a functionality for rule of PCM substream parameters.
Typically, when userspace opens PCM character devices, each driver adds
its own rules to PCM substream according to design of hardware. When the
userspace executes hw_params ioctl with favorite parameters, the actual
parameters are calculated according to the rules and the given parameters.
Then, the result is returned to userspace.
Currently, ALSA Dice driver has the rule between channels and rates, while
Dice interface design doesn't allow drivers to retrieve all of the
combinations. Dice drivers are just allowed to get current sampling
transfer frequency and the number of multi bit linear audio data channels
in an data block of an AMDTP packet.
This commit purges the rule, and limit PCM substreams to current sampling
transfer frequency, following to the interface design.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/firewire/digi00x/amdtp-dot.c:67: warning: type qualifiers ignored on function return type
Drop the bogus "const" type qualifier on the return type of dot_scrt()
to fix this.
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 3beab0f844 added a member for control and status message, while
it's planned and not implemented yet.
This commit removes it.
Fixes: 3beab0f844fa('ALSA: firewire-tascam: add support for outgoing MIDI messages by asynchronous transaction')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, 'struct snd_tscm_spec' has a member named as 'is_controller' to
identify MIDI controller. This member was originally added to skip
parse control and status messages in isochronous packets for non-controller
model.
As long as I investigate, FW-1804 (non-controller) also transfers the
control and status message, thus it becomes meaningless.
This commit removes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This model supports:
* maximum 12 PCM channels for PCM playback
* maximum 18 PCM channels for PCM capture
* 4 ports for MIDI playback
* 4 ports for MIDI capture
* control and status messages in tx isochronous packets
* up to 96.0 kHz
This commit adds support for the model. As the other supported models,
all of available PCM channels are always enabled.
As I described in commit c0949b2785, Ilya Zimnovich had investigated
TASCAM FireWire series in 2011 with his FW-1804. In his report, this model
has internal multiplexer and any software implementation can control it.
Following to the design of ALSA firewire stack, this commit won't
implement it. It should be in userspace via Linux fw character device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When unsupported models are connected, snd-firewire-tascam module causes
NULL pointer dereference in fw_core_remove_address_handler() (due to
list_del_rcu()).
This commit prevents this bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return type "unsigned int" was used by the get_formation_index function
despite of the aspect that it will eventually return a negative error code.
So, change to signed int and get index by reference in the parameters.
Done with the help of Coccinelle.
[Fix the missing braces suggested by Julia Lawall -- tiwai]
Signed-off-by: Lucas Tanure <tanure@linux.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some users have reported that their Dice based models generate ETIMEDOUT
when starting PCM playback. It means that current timeout (=100msec) is
not enough for their models to transfer notifications.
This commit expands the timeout up to 2 sec. As a result, in a worst case,
any operations to start AMDTP streams takes 2 sec or more. Then, in
userspace, snd_pcm_hw_params(), snd_pcm_prepare(), snd_pcm_recover(),
snd_rawmidi_open(), snd_seq_connect_from() and snd_seq_connect_to() may
take the time.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, card registration is processed under situation
with few bus reset. There's no need to add a workaround of transaction
re-initialization at timeout.
This commit purges the re-initialization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models based on ASIC for Dice II series (STD, CP) change their
hardware configurations after appearing on IEEE 1394 bus. This is due to
interactions of boot loader (RedBoot), firmwares (eCos) and vendor's
configurations. This causes current ALSA dice driver to get wrong
information about the hardware's capability because its probe function
runs just after detecting unit of the model.
As long as I investigated, it takes a bit time (less than 1 second) to load
the firmware after bootstrap. Just after loaded, the driver can get
information about the unit. Then the hardware is initialized according to
vendor's configurations. After, the got information becomes wrong.
Between bootstrap, firmware loading and post configuration, some bus resets
are observed.
This commit offloads most processing of probe function into workqueue and
schedules the workqueue after successive bus resets. This has an effect to
get correct hardware information and avoid involvement to bus reset storm.
For code simplicity, this change effects all of Dice-based models, i.e.
Dice II, Dice Jr., Dice Mini and Dice III.
I use a loose strategy to manage a race condition between the work and the
bus reset. This is due to a specification of dice transaction. When bus
reset occurs, registered address for the transaction is cleared. Drivers
must re-register their own address again. While, this operation is required
for the work because the work includes to wait for the transaction. This
commit uses no lock primitives for the race condition. Instead, checking
'registered' member of 'struct snd_dice' avoid executing the work again.
If sound card is not registered, the work can be scheduled again by bus
reset handler.
When .remove callback is executed, the sound card is going to be released.
The work should not be pending or executed in the releasing. This commit
uses cancel_delayed_work_sync() in .remove callback and wait till the
pending work finished. After .remove callback, .update callback is not
executed, therefore no works are scheduled again.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before allocating an instance of sound card, ALSA dice driver checks
chip_ID_hi in Bus information block of Config ROM, then also checks
subaddresses. The former operation reads cache of Config ROM in Linux
FireWire subsystem, while the latter operation sends read transaction.
The latter can be merged into initialization of transaction system.
This commit splits these two operations to reduce needless transactions
in probe processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I investigate SCS.1m, this model reports to transfer/receive
PCM data channels/MIDI conformant data channels in tx/rx AMDTP packet.
There's a contradiction that this model actually has no analog/digital
capture port for PCM frames and no physical MIDI ports.
I guess that SCS.1d also has the contradiction. This model has no
analog/digital ports for PCM frames and no physical MIDI ports, thus it
requires no streaming functionality.
This commit adds some modification codes to handle the contradiction,
as much as possible. Unfortunately, this module adds one PCM playback
substream for SCS.1d so as SCS.1m.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now ALSA oxfw driver gains functionalities which scs1x module has.
This commit obsoletes the scs1x module, and adds a line of MODULE_ALIAS
to load oxfw module instead of scs1x module.
In scs1x module, the name of 'shortname' field is fixed as 'SCS1x'. This
field is used to name MIDI ports for both of SCS.1m and SCS.1d. This is
not good because typically some SCS.1m and SCS.1d are used in the same
system. It's better to distinguish them according to name of the ports.
This commit applies model name in config ROM to the 'shortname'.
For the name of 'driver' and 'longname', this commit uses the same way
applied to the other models. This change may not bring disadvantages to
users because userspace applications use ALSA rawmidi or seq interface
and these interfaces are not influenced by them directly.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit copies some functions of asynchronous transactions for MIDI
playback, to merge scs1x module. The features of payload in asynchronous
transaction are the same as captured MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit copies some functions of asynchronous transactions for MIDI
capture, to merge scs1x module. The features of payload in asynchronous
transaction are:
* System exclusive messages for SCS.1 are encoded without ID data. In
this encoding scheme, 4 bits in LSB are available. The bits are squashed
in payload byte. Thus, one payload byte transfers two MIDI messages.
* The first byte of payload byte means:
* 0x00: depending on second payload byte
* 0xf9: including escaped system exclusive messages for SCS.1, up to
3 byte (= 6 MIDI messages)
* the others: including MIDI 1.0 messages
* the others: including escaped system exclusive messages for SCS.1, up
to 64 bytes
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When physical controls on SCS.1 models are operated, the models transfer
MIDI messages in asynchronous transactions on IEEE 1394 bus. The models
have a register to have an address for the transactions, and drivers
can register own address for this purpose.
This commit keeps a region of address, registers it and adds a handler for
the transactions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanton Controllers and Systems 1 (SCS.1) series is supported by ALSA
scs1x driver. This driver just supports MIDI functionality. On the other
hand, models in this series are based on OXFW971 and ALSA OXFW driver can
support them.
SCS.1 series has MIDI functionality to control its surface state such as
LED lighting. When operating physical knobs and faders, the models
generate MIDI messages. These MIDI messages are transferred by asynchronous
transactions. These transactions are really model-specific and ALSA OXFW
driver requires the functionality so as scs1x module implements.
This commit adds scs1x layer as a preparation to merge scs1x driver to
oxfw driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, some model-specific members are split from the
structure. The structure is just to keep names for compatibility to old
drivers.
This commit arranges name of the structure and localize it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, some members are moved from 'struct snd_oxfw' because
they're model-specific. There are also the other model-specific parameters
in 'struct device_info'.
This commit moves these members to model-specific structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, 'struct snd_oxfw' has some members for models supported by old
firewire-speakers driver, while these members are useless to the other
models.
This commit allocates new memory block and moves these members to
model-specific structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA oxfw driver should have backward compatibility to old
firewire-speakers driver. Additionally, in future commit, scs1x driver
will be merged. It's nice to add a pointer to have a memory block for
model-specific structures.
This commit adds a member to 'struct snd_oxfw' for this aim. Deallocation
is done at freeing ALSA card structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding control elements is just for models supported by old
firewire-speakers modules. The processing should be in a function to add
model-dependent quirk.
This commit moves the codes to the function. As a result, the function
should handle error state, thus this commit also changes prototype of
the function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, assignment to model-dependent quirk is corresponding to
asynchronous transactions on IEEE 1394 bus. This is also achieved with
device entry.
This commit changes the processing of model-dependent quirk with the
entry. As a result, the transactions are sent only for Loud models.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA OXFW driver uses AV/C Audio Subunit commands to control some models.
The commands get/set the state of Feature function block of the subunit.
The commands are not specific to OXFW, thus there's a possibility to use
them in the other drivers.
Currently, helper functions for the commands require 'struct snd_oxfw',
although, it's not necessarily required. It's better to change prototype
of the functions without the structure for future use.
This commit changes the prototype.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames local functions with prefix 'spkr_', so that they're
for firewire-speakers.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA firewire stack, drivers basically has no control elements. This
is due to the fact that each model has own functionality even if they use
the same communication chipset. Implementing all of the functionalities in
kernel space unreasonably increases our efforts to maintain the stack. In
most case, these functionalities can be implemented in userspace via Linux
fw character devices.
However, ALSA OXFW driver has control elements comes from old
firewire-speakers driver. Adding the elements is in a file names as
'oxfw-control.c', while the elements are really model-specific. The
name is confusing because it gives an idea to handle control elements
for all of OXFW-based models.
This commit renames the file so that it's just for models supported by
old firewire-speakers driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The counter is incremented/decremented in critical section protected with
mutex. Therefore, no need to use atomic_t.
This commit changes the type to unsigned int.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, critical section is protected by mutex in functions of
fireworks_stream.c. Callers increments/decrements substreams counter
before calling the functions. Moving mutex to the callers code allows
to change type of the substeram counter from atomic_t to unsigned int.
This commit is a preparation for obsoleting usage of atomic_t for
substream counter.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit a471fcde8c ("ALSA: dice: fix detection of Weiss devices") adds
a quirk of Weiss models. According to users' reports, Loud models also
have the similar quirk. They have 0x10 in the category field.
This commit adds support for Mackie Onyx Blackbird and Onyx-i series.
As long as I know, Dice-based models produced by
Focusrite/Alesis/PreSonus/M-Audio/TC Electronic have default value (0x04)
in their category field, thus it may be reasonable to add a condition
statement for Loud models, instead of removing the check of category value.
Reported-by: Rouge Etienne <erouge.externe@m6.fr>
Reported-by: Etilem <contact@etilem.net>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A commit to add support for this model should have added a comment
about this model to Kconfig.
Fixes: 759a2f40c9fa('ALSA: oxfw: add an entry for TASCAM FireOne')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family uses two ways to transfer MIDI messages. They're
different mechanisms, while it's better to handle the ways in different
ALSA rawmidi character devices because one character device has just a
set of operations.
This commit adds another rawmidi character device for control MIDI port. As
a result, first rawmidi character device is just for MIDI messages
transferred by isochronous packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family has two types of MIDI port; one is for physical MIDI
port and another is for MIDI control message. The former is transferred in
isochronous packet, and the latter is transferred by asynchronous
transaction. These transmission mechanisms are completely different, while
current ALSA digi00x driver defines a set of operations for them with
several condition statements. As a result, codes for the operation are
messy.
This commit adds a set of MIDI operation for control MIDI ports. In later
commit, it's applied as an operation for ALSA rawmidi character device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In following commit, new functions and variables are added for operations
of MIDI control port.
This commit is a preparation. Current identifiers are renamed so that they
mean physical MIDI ports.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When MIDI buffer stores two or more MIDI messages, TASCAM driver
transfers asynchronous transactions including one MIDI message and
extra bytes from second MIDI message.
This commit fixes this bug by clearing needless bytes in the buffer. The
consumed bytes are already calculated correctly, thus the sequence of
transactions is already correct.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In transactions for MIDI messages, the first byte is used for label and
the rest is for MIDI bytes. In current code, these are handled correctly,
while there's a small mistake for loop condition to include meaningless
statement.
This commit adds two local variables for them and improve the loop
condition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the callback function of asynchronous MIDI port, the intension of some
local variables are not clear.
This commit improves them. The 'len' variable is used to calculate the
number of MIDI bytes including in the transaction. The 'consume' variable
is used to return the actual number of consumed bytes in ALSA MIDI buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the callback function of asynchronous MIDI port, some local variables
are declared 'unsigned int', while they're assigned to int value of return
from snd_rawmidi_transmit_peek().
This commit fixes the type.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The given buffer to callback function is cleared in caller side.
This commit removes buffer initialization in callee side.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's leave space for the NUL char otherwise the static checkers
complain that we go beyond the end of the array.
Fixes: 53b3ffee78 ('ALSA: firewire-tascam: change device probing processing')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireOne is based on OXFW971 and ALSA OXFW driver can support it.
These are values of identical registers.
$ ./firewire-request /dev/fw1 read 0xfffff0050000
result: 97100105
$ ./firewire-request /dev/fw1 read 0xfffff0090020
result: 39373100
This commit adds an entry for this model. This model has physical controls
and its MIDI control messages are transferred to second MIDI data stream
multiplexed in one MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, sequence multiplexing is applied to MIDI conformant data
channel. As a result, eight MIDI data streams are included in the channel.
Although ALSA AM824 data block processing layer implements this
multiplexing, current OXFW driver doesn't utilize it due to wrong
calculation of MIDI ports.
This commit fixes this bug to add proper calculation. Although this commit
allows to use 8 MIDI data streams, the number of available MIDI ports is
limited by the number of ALSA MIDI ports added by the driver.
Fixes: df075feefbd3('ALSA: firewire-lib: complete AM824 data block processing layer')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current OXFW driver calculates the number of MIDI ports just before adding
ALSA MIDI ports. It's convenient for some devices with quirks to move
these codes before handling quirks.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When asynchronous MIDI port is closed before callbacked, the callback
function causes NULL pointer dereference to missing MIDI substream.
This commit fixes this bug.
Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The contents of Config ROM in firewire device structure are already
aligned to CPU-endianness. Thus, no need to convert it again.
This commit removes needless conversions
Fixes: 9edf723fd858('ALSA: firewire-digi00x: add skeleton for Digi 002/003 family')
Fixes: c0949b278515('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, metering is supported for BeBoB based models
customized by M-Audio. The data in transaction is aligned to
big-endianness, while in the driver code u16 typed variable is assigned
to the data. This causes sparse warnings.
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
This commit fixes this bug by using __be16 variable for the data.
Fixes: 3149ac489ff8('ALSA: bebob: Add support for M-Audio special Firewire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, snd_efw_command_get_phys_meters() was added to handle
metering data. The given buffer is used to save transaction result and to
convert between endianness. But this causes sparse warnings.
fireworks_command.c:269:25: warning: incorrect type in argument 1 (different base types)
fireworks_command.c:269:25: expected unsigned int [usertype] *p
fireworks_command.c:269:25: got restricted __be32 [usertype] *
This commit fixes this bug.
Fixes: bde8a8f23bbe('ALSA: fireworks: Add transaction and some commands')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, u32 data was assigned to __be32 variable instead of an
int variable. This is not enough solution because it still causes sparse
warnings.
dice.c:80:23: warning: incorrect type in assignment (different base types)
dice.c:80:23: expected restricted __be32 [usertype] value
dice.c:80:23: got unsigned int
dice.c:81:21: warning: restricted __be32 degrades to integer
dice.c:81:46: warning: restricted __be32 degrades to integer
This commit fixes this bug.
Fixes: 7c2d4c0cf5ba('ALSA: dice: Split transaction functionality into a file')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some local variables in some functions are typed as unsigned int, while
__be32 value is assigned to them. This causes sparse warnings.
dice-stream.c:50:17: warning: incorrect type in assignment (different base types)
dice-stream.c:50:17: expected unsigned int [unsigned] channel
dice-stream.c:50:17: got restricted __be32 [usertype] <noident>
dice-stream.c:74:17: warning: incorrect type in assignment (different base types)
dice-stream.c:74:17: expected unsigned int [unsigned] channel
dice-stream.c:74:17: got restricted __be32 [usertype] <noident>
This commit fixes this bug.
Fixes: 288a8d0cb04f('ALSA: dice: Change the way to start stream')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A former commit moves oxfw-related codes to a sub-directory, while it
forgot to remove an entry from Makefile in parent directory.
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When committed to upstream, these four modules had wrong entries for
Makefile. This forces them to be loadable modules even if they're set
as built-in.
This commit fixes this bug.
Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices')
Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices')
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My static checker complains because tscm->spec->midi_capture_ports is
either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so
this is possibly off by one. I have looked at the code and I think it
should be >= instead of > as well.
Fixes: 107cc0129a ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We recently tried to add some new code to support turning the LED on and
off but the code in snd_tscm_transaction_reregister() is unreachable.
Fixes: e65e2cb99e ('ALSA: firewire-tascam: Turn on/off FireWire LED')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, this driver picks up model name with be32_to_cpu() macro
to align characters. This is wrong operation because the result is
different depending on CPU endiannness.
Additionally, vendor released several versions of firmware for this
series. It's not better to assign model-dependent information to
device entry according to the version field.
This commit fixes these bugs. The name of model is picked up correctly
and used to identify model-dependent information.
Cc: Stefan Richter <stefanr@s5r6.in-berlin.de>
Fixes: c0949b2785 ('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has some LEDs on its surface. These LEDs can be
turned on/off by receiving asynchronous transactions to a certain
address. One of the LEDs is labels as 'FireWire'. It's better to light it
up when this driver starts to work. Besides, the LED for 'FireWire' is
turned off at bus reset.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, this driver got functionalities to transfer/receive
MIDI messages to/from TASCAM FireWire series.
This commit adds some ALSA MIDI ports to enable userspace applications
to use the functionalities.
I note that this commit doesn't support virtual MIDI ports which console
models support. A physical controls can be assigned to a certain MIDI
ports including physical and virtual. But the way is not clear.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to receive MIDI
messages. The transaction should be sent to a certain address.
This commit supports the outgoing MIDI messages. The messages in the
transaction includes some quirks:
* One MIDI message is transferred in one quadlet transaction, except for
system exclusives.
* MIDI running status is not allowed, thus transactions always include
status byte.
* The basic data format is the same as transferring MIDI messages
supported in previous commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to transfer MIDI
messages. The transaction is sent to a registered address.
This commit supports the incoming MIDI messages. The messages in the
transaction include some quirks:
* Two quadlets are used for one MIDI message and one timestamp.
* Usually, the first byte of the first quadlet includes MIDI port and MSB
4 bit of MIDI status. For system exclusive message, the first byte
includes MIDI port and 0x04, or 0x07 in the end of the message.
* The rest of the first quadlet includes MIDI bytes up to 3.
* Several set of MIDI messages and timestamp can be transferred in one
block transaction, up to 8 sets.
I note that TASCAM FireWire series ignores ID bytes of system exclusive
message. When receiving system exclusive messages with ID bytes on physical
MIDI bus, the series transfers the messages without ID bytes on IEEE 1394
bus, and vice versa.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, asynchronous transactions are supported for physical
controls. This commit adds a pair of MIDI ports for them.
This driver already adds diferrent number of ALSA MIDI ports for physical
MIDI ports, and the number of in/out ports are different. As seeing as
'amidi' program in alsa-utils package, a pair of in/out MIDI ports is
expected with the same name. Therefore, this commit adds a pair of new
ports to the first.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, asynchronous transaction for incoming MIDI messages
from physical controls is supported. The physical controls may be
controlled by receiving MIDI messages at a certain address.
This commit supports asynchronous transaction for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 00x series has two types of model; rack and console. The console
models have physical controls. The model can transmit control messages.
These control messages are transferred by asynchronous transactions to
registered address.
This commit supports the asynchronous transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds MIDI functionality to capture/playback MIDI messages
from/to physical MIDI ports. These messages are transferred in isochronous
packets.
When no substreams request AMDTP streams to run, this driver starts the
streams at current sampling rate. When other substreams start at different
sampling rate, the streams are stopped temporarily, then start again at
requested sampling rate. This operation can generate missing MIDI bytes,
thus it's preferable to start PCM substreams at favorite sampling rate in
advance.
Digi 002/003 console also has a set of MIDI port for physical controls.
These ports are added in later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Digi 002/003 protocol, MIDI messages are transferred in the last data
channel of data blocks. Although this data channel has a label of 0x80,
it's not fully MIDI conformant data channel especially because the Counter
field always zero independently of included MIDI bytes. The 4th byte of
the data channel in LSB tells the number of included MIDI bytes. This byte
also includes the number of MIDI port. Therefore, the data format in this
data channel is:
* 1st: 0x80 as label
* 2nd: MIDI bytes
* 3rd: 0 or MIDI bytes
* 4th: the number of MIDI byte and the number of MIDI port
This commit adds support of MIDI messages in data block processing layer.
Like AM824 data format, this data channel has a capability to transfer
more MIDI messages than the capability of phisical MIDI bus. Therefore, a
throttle for data rate is required to prevent devices' internal buffer to
overflow.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original code for 'DoubleOhThree' encoding was written with '__u8' type,
while the type is usually used to export something to userspace.
This commit replaces the type with 'u8'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In firewire-lib, isochronous packet streaming is stopped when detecting
wrong value for FMT field of CIP headers. Although this is appropriate
to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization
use invalid value to FMT field of CIP headers in the beginning of
streaming.
$ journalctl
snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff
I got this log with M-Audio FireWire 1814. In this line, the value of FMT
field is 0x00, while it should be 0x10 in usual AMDTP.
Except for the beginning, these devices continue to transfer packets with
valid value for FMT field, except for the beginning. Therefore, in this
case, firewire-lib should continue to process packets. The former
implementation of firewire-lib performs it.
This commit loosens the handling of wrong value, to continue packet
processing in the case.
Fixes: 414ba022a5 ('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec,
snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are
initialized. Make them all const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.
This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.
This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.
This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.
This commit serialize request/response transactions, by adding one
boolean member to represent idling state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.
This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.
The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds hwdep interface so as the other IEEE 1394 sound devices
has.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
When one of PCM substreams are running or external clock source is
selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds streaming functionality for both direction. To utilize
the sequence of the number of data blocks in packets, full duplex with
synchronization is applied.
Besides, TASCAM FireWire series allows drivers to decide which PCM data
channels are enabled. For convenience, this driver always enable whole the
data channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series uses non-blocking transmission for AMDTP packet
streaming, while the format of data blocks is unique.
The CIP headers includes specific value in FMT field and no SYT
information.
In transmitted packets, the first data channel represents event counter,
and the last data channel has status and control information. The rest
has 24bit PCM samples with right padding.
In received packets, all of data channels include 16, 24, 32bit PCM
samples. There's no other kind of information.
This commit adds support for this protocol. For convenience, the size of
PCM samples in outgoing packet is limited by 16 and 24bit. The status and
control information will be supported in future commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has certain registers for firmware information.
This commit adds proc node to show the information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series doesn't tell drivers their capabilities, thus
the drivers should have model-dependent parameters and apply it to
detected devices.
This commit adds a structure to represent such parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for TASCAM FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.
TASCAM FireWire series consists of:
* PDI 1394P23 for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* XILINX XC9536XL
* XILINX Spartan-II XC2S100
* ATMEL AT91M42800A
Ilya Zimnovich had investigated TASCAM FireWire series in 2011, and
discover some features of his FW-1804. You can see a part of his research
in FFADO project.
http://subversion.ffado.org/wiki/Tascam
A part of my work are based on Ilya's investigation, while this series
doesn't support the FW-1804, because of a lack of config ROM
information and its protocol detail, especially for PCM channels.
I observed that FW-1884 and FW-1082 don't work properly with 1394 OHCI
controller based on VT6315. The controller can actually communicate packets
to these models, while these models generate no sounds. It may be due to
the PHY/LINK layer issues. Using 1394 OHCI controller produced by the other
vendors such as Texas Instruments may work. Or adding another node on the
bus.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family uses asynchronous transaction for messaging.
The address to transmit this message is stored on a certain register.
This commit allocates a range of address on OHCI 1394 host controller
to handle the messaging. As long as I know, the purpose of this message
seems to notify lost of synchronization. While, the meaning of content
of the message is not clear.
Actual examples of this messaging:
* When clock source is set as internal:
- 0x00007051
- 0x00007052
- 0x00007054
- 0x00007057
- 0x00007058
* When clock source is set as somewhat external:
- 0x00009000
- 0x00009010
- 0x00009020
- 0x00009021
- 0x00009022
The lost often occurs when using internal clock source. In this case,
users hear sounds with quite short gap every several minutes. In fact,
the lost is recovered temporarily.
When using with external clock source, the lost seems not to occur. The
mechanism is not clear yet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds hwdep interface so as the other sound drivers for units
on IEEE 1394 bus have.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
Any PCM substreams are jointed because incoming/outgoing AMDTP streams
are bound. When one of PCM substream is running or external clock source
is selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc node to show current clock status for debugging.
As long as testing Digi 002 rack, registers can show local clock rate,
local clock source. When external clock input such as S/PDIF is
connected, the registers show the detection and external clock rate.
Additionally, the registers show the mode of optical digital input
interface. Although, a tester with Digi 003 rack reports this makes no
sense. Further investigation is required for Digi 003 series.
Besides, in Digi 002 rack, the S/PDIF format must be IEC 60958-4,
so-called professional.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to manage streaming.
The streaming is not controlled by CMP in IEC 61883-6. It's controlled by
IEEE 1394 write transaction to certain addresses.
Several clock sources are available, while there're no differences about
packet transmission. The value of SYT field in transmitted packets is
always zero. Thus, streams in both direction don't build synchronization.
And the device always requires received packets to transmit packets. This
driver keeps to transfer outgoing stream even if they're not required.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family uses its own format for data blocks. The format is
quite similar to AM824 in IEC 61883-6, while there're some differences:
* The Valid Bit Length (VBL) code is always 0x40 in Multi-bit Linear Audio
(MBLA) data channel.
* The first data channel includes MIDI messages, against IEC 61883-6
recommendation.
* The Counter field is always zero in MIDI conformant data channel.
* Sequence multiplexing in IEC 61883-6 is not applied to the MIDI
conformant data channel.
* PCM samples are scrambled in received AMDTP packets. We call the way
as Double-Oh-Three (DOT). The algorithm was discovered by
Robin Gareus and Damien Zammit in 2012.
This commit adds data processing layer to satisfy these differences.
There's a quirk about transmission mode for received packets. When this
driver applies non-blocking mode to outgoing packets with isochronous
channel 2 or more, after 15 to 20 seconds since playbacking, any PCM
samples causes noisy sound on the device. With isochronous channel 0 or 1,
this doesn't occur. As long as I investigated, this quirk is not observed
when applying blocking mode to the received packets.
This driver applies blocking mode to outgoing packets, while non-blocking
mode to incoming packgets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for Digidesign 002/003 family. This commit
just creates/removes card instance according to bus event. More functions
will be added in following commits.
Digidesign 002/003 family consists of:
* Agere FW802B for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* ALTERA ACEX EP1K50 for IEC 61883 layer and DSP controller
* ADSP-21065L for signal processing
[minor cleanup using skip_spaces() by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit moves the codes related to data block processing from packet
streaming layer to AM824 layer.
Each driver initializes amdtp stream structure for AM824 data block by
calling amdtp_am824_init(). Then, a memory block is allocated for AM824
specific structure. This memory block is released by calling
amdtp_stream_destroy().
When setting streaming parameters, it calls amdtp_am824_set_parameters().
When starting packet streaming, it calls amdtp_stream_start(). When
stopping packet streaming, it calls amdtp_stream_stop().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.
This commit renames PCM format setting function to prepare for integrating
AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.
This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.
Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.
In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.
This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.
This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.
In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.
This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.
Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.
This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.
This commit support to set arbitrary values for the fields.
In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.
Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.
Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.
This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.
The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.
This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.
On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee
("ALSA: dice: Split stream functionality into a file").
Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.
In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.
For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.
This commit changes this function to return error code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mackie Onyx Satellite has two usage; standalone and with base station.
These two modes has different stream formats. In standalone mode, rx data
block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data
block includes 2. With base station, they're 6 and 2.
Although, with base station, the actual tx packet include wrong value in
'dbs' field in its CIP header. This quirk causes packet streaming layer to
detect packet discontinuity and to stop PCM substream.
This is a response of 'single' subfunction 'extended stream format
information' command in AV/C Stream Format Information Specification 1.1.
It means that a data block in tx packets includes 2 MBLA data channels.
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff
response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02
response: 010: 06
Current OXFW driver parses the response and detects stream formats
correctly.
$ cat /proc/asound/card1/firewire/formation
...
Output Stream from device:
Rate PCM MIDI
* 48000 2 0
44100 2 0
88200 2 0
96000 2 0
On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number
of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of
two successive payload quadlets, it should be multiple of 2.
payload CIP CIP
quadlets header0 header1
24 00060052 9002ffff
24 0006005e 9002ffff
26 0006006a 9002ffff
24 00060077 9002ffff
24 00060083 9002ffff
26 0006008f 9002ffff
24 0006009c 9002ffff
24 000600a8 9002ffff
26 000600b4 9002ffff
24 000600c1 9002ffff
This commit adds support for this quirk to OXFW driver, by using
CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses
the value of its 'data_block_quadlets' member instead of the value in CIP
header. This value is already set by OXFW driver and no discontinuity is
detected.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In PCM core, when hw_params() in each driver returns error, the state of
PCM substream is kept as 'open'. In this case, current drivers for sound
units on IEEE 1394 bus doesn't decrement substream counter in hw_free()
correctly. This causes these drivers to keep streams even if not
required.
This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer()
fails, hw_params function in each driver returns without incrementing the
counter.
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on from it.
Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.
As long as I investigated, AudioFire2/AudioFire4/AudioFirePre8 have a
quirk to transfer a first packet with 0x02 in its dbc field. This causes
ALSA Fireworks driver to detect discontinuity. In this case, firmware
version 5.7.0, 5.7.3 and 5.8.0 are used.
Payload CIP CIP
quadlets header1 header2
02 00050002 90ffffff <-
42 0005000a 90013000
42 00050012 90014400
42 0005001a 90015800
02 0005001a 90ffffff
42 00050022 90019000
42 0005002a 9001a400
42 00050032 9001b800
02 00050032 90ffffff
42 0005003a 9001d000
42 00050042 9001e400
42 0005004a 9001f800
02 0005004a 90ffffff
(AudioFire2 with firmware version 5.7.)
$ dmesg
snd-fireworks fw1.0: Detect discontinuity of CIP: 00 02
These models, AudioFire8 (since Jul 2009 ) and Gibson Robot Interface
Pack series uses the same ARM binary as their firmware. Thus, this
quirk may be observed among them.
This commit adds a new member for AMDTP structure. This member represents
the value of dbc field in a first AMDTP packet. Drivers can set it with
a preferred value according to model's quirk.
Tested-by: Johannes Oertei <johannes.oertel@uni-due.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9c6893e0be.
The fix is superseded by the next commit as a better implementation
for supporting AudioFire2/AudioFire4/AudioFirePre8 quirks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on.
Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.
AudioFire2 has a quirk to transfer a first packet with non-zero in
its dbc field. This causes ALSA Fireworks driver to detect discontinuity.
As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk.
This commit adds a support for the quirk to handle AudioFire2 packets.
For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of
AudioFire2's firmwares.
02 00050002 90ffffff <-
42 0005000a 90013000
42 00050012 90014400
42 0005001a 90015800
02 0005001a 90ffffff
42 00050022 90019000
42 0005002a 9001a400
42 00050032 9001b800
02 00050032 90ffffff
42 0005003a 9001d000
42 00050042 9001e400
42 0005004a 9001f800
02 0005004a 90ffffff
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.
For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
- snd_pcm_prepare()
- snd_pcm_hw_params()
- snd_pcm_recover()
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BeBoB installed devices have BeBoB register area. This area stores
basic information about its firmware. A register has its protocol
version.
This commit adds 'version' member and store the device's protocol
version to handle v3 quirks in following commits.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.
This commit purge the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The old string literals were completely replaced by new normalized
representation.
This commit obsoletes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.
This commit is a preparation for replacement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous commit allows this driver to detect several types of clock
source, while there's no normalized expression for it.
This commit adds a new enumerator for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
When detecting packet discontinuity, handle_in_packet() returns minus value
and this value is assigned to unsigned int variable, then the variable has
huge value. As a result, the variable causes buffer-over-run in
handle_out_packet(). This brings invalid page request and system hangup.
This commit fixes the bug to add a new argument into handle_in_packet()
and the number of handled data blocks is assignd to it. The function
return value is just used to check error.
I also considered to change the type of local variable to 'int' in
in_stream_callback(). This idea is based on type-conversion in C standard,
while it may cause future problems when adding more works. Thus, I dropped
this idea.
Fixes: 6fc6b9ce41c6('ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device is based on DM1000E, and BeBoB version 1 firmware is
installed.
$ cat /proc/asound/cards
0 [Pro ]: BeBoB - Mbox 2 Pro
DIGIDESIGN Mbox 2 Pro (id:1, rev:1),
GUID 00a07e0100a90000 at fw1.0, S400
$ cat /proc/asound/Pro/firewire/firmware
Manufacturer: bridgeCo
Protocol Ver: 1
Build Ver: 0
GUID: 0x00A07E0100A90000
Model ID: 0x01
Model Rev: 1
Firmware Date: 20071031
Firmware Time: 034402
Firmware ID: 0xA9
Firmware Ver: 16777215
Base Addr: 0x20080000
Max Size: 1572864
Loader Date: 20051207
Loader Time: 205554
With this patch, ALSA BeBoB driver can start packet streaming to/from
this model, while as a default, internal multiplexer of this model is
not initialized and generates no sound even if the driver transfers
any packets with PCM samples. To hear any sounds from this model,
userspace applications should be developed to set parameters to the
internal multiplexer. You can see raw information in FFADO website:
http://subversion.ffado.org/wiki/AvcModels/DigiDesignMboxPro2
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When detecting zero in 'dbs' field of CIP header, this packet streaming
should be aborted because of avoiding division-by-zero. This is an error
in an aspect of IEC 61883-1, thus protocol error.
This commit use EPROTO instead of EIO. Actually, the returned value is
not used for userspace and this commit has no effect.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When detecting invalid value in 'dbs' field of CIP header or packet
discontinuity, current implementation reports the status by err_info().
In most cases this state is caused by model-specific issue due to
vendor's customization and should be reported to developers.
This commit use dev_err() instead of dev_info() for this purpose.
In the cases, packet streaming is aborted, thus no message floading
occurs.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some macros include my misunderstanding for IEC 61883-1 or -6.
Additionally, some fixed values appear on codes.
This commit replaces these with macros with proper names.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The naming rule for local functions was inconsistent. This commit
rename them with a consistent manner.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Former patches allow non-blocking streams to synchronize with timestamp.
This patch removes the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, error handling for incoming packet processing is
outside of packetization. This is nice for reading the codes.
This commit applies this idea for outgoing packet processing, too.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current implementation reuses the value of syt field in incoming packet to
outgoing packet for full duplex timestamp synchronization, while the number
of data blocks in outgoing packets refers to hard-coded table and the
synchronization cannot be applied to non-blocking stream.
This commit passes the number of data blocks from incoming packet
processing to outgoing packet processing for the synchronization. For
normal mode, isochronous callback handler is changed to generate the values
of syt and data blocks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function is called according to conditions between the value of
syt and streaming mode(blocking or non-blocking).
To simplify caller's work, this commit push these conditions to the
function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, the number of data blocks in a packet is limited up to
the value of SYT_INTERVAL. Current implementation is compliant to the
limitation, while it can cause buffer-over-run when the value of dbs
field in received packet is illegally large.
This commit adds a validator to detect such illegal packets to prevent
the buffer-over-run. Actually, the buffer is aligned to the size of memory
page, thus this issue hardly causes system errors due to the room to page
alignment, as long as a few packets includes such jumbo payload; i.e.
a packet to several received packets.
Here, Behringer F-Control Audio 202 (based on OXFW 960) has a quirk to
postpone transferring isochronous packet till finish handling any
asynchronous packets. In this case, this model is lazy, transfers no
packets according to several cycle-start packets. After finishing, this
model pushes required data in next isochronous packet. As a result, the
packet include more data blocks than IEC 61883-6 defines.
To continue to support this model, this commit adds a new flag to extend
the length of calculated payload. This flag allows the size of payload
5 times as large as IEC 61883-6 defines. As a result, packets from this
model passed the validator successfully.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some M-Audio devices require to receive bootup command just after
powering on, while codes in BeBoB driver doesn't work properly in
big-endian machine because the command should be aligned by
little-endian.
This commit fixes this bug. This fix should go to stable kernel.
Cc: Takayuki Shiroma <t.shiroma.oki@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With previous commit, this module managed to leave the counting to each
drivers, but the isochronous resources functionality still increment/decrement
the count.
This commit purge such codes to leave the responsibility to each drivers.
Fix: c6f224dc20 ('ALSA: firewire-lib: remove reference counting')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For received packet stream, the offset of 'RX_SEQ_START' locates after
the offset of 'RX_NUMBER_MIDI', although current macro and proc output
includes wrong offsets.
Fortunately, this bug doesn't affect streaming functionality because
these macro is not used.
This commit fixes these wrong macro and outputs.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amdtp_stream_wait_callback() doesn't return minus value and
the return code is not for error code.
This commit fixes with a propper condition and an error code.
Fixes: f3699e2c77 ('ALSA: oxfw: Change the way to start stream')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A part of these drivers, especially BeBoB driver, are programmed to wait
some events. Thus the drivers should not destroy any data in .remove()
context.
This commit moves some destructors from 'struct fw_driver.remove()' to
'struct snd_card.private_free()' to shutdown safely.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently stream destructor in each driver has a problem to be called in
a context in which sound card object is released, because the destructors
call amdtp_stream_pcm_abort() and touch PCM runtime data.
The PCM runtime data is destroyed in application's context with
snd_pcm_close(), on the other hand PCM substream data is destroyed after
sound card object is released, in most case after all of ALSA character
devices are released. When PCM runtime is destroyed and PCM substream is
remained, amdtp_stream_pcm_abort() touches PCM runtime data and causes
Null-pointer-dereference.
This commit changes stream destructors and allows each driver to call
it after releasing runtime.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AMDTP helper functions increment/decrement reference counter for an
instance of FireWire unit, while it's complicated for each driver to
process error state.
In previous commit, each driver has the role of reference counting. This
commit removes this role from the helper function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks and Dice drivers try to touch instances of FireWire unit after
sound card object is released, while references to the unit is decremented
in .remove(). When unplugging during streaming, sound card object is
released after .remove(), thus Fireworks and Dice drivers causes GPF or
Null-pointer-dereferencing to application processes because an instance of
FireWire unit was already released.
This commit adds reference-counting for FireWire unit in drivers to allow
them to touch an instance of FireWire unit after .remove(). In most case,
any operations after .remove() may be failed safely.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5'
by UTF-8 character encoding scheme. But the byte sequence was converted
to '0x c3 82 c2 b5' in a previous commit. As a result, the byte
sequence cannot represent microsecond sign in UTF-8 or ASCII. This
may confuse developers.
This commit replaces the sign to string expression with 'microseconds'
to purge superfluous troubles.
Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do no send MIDI bytes at the full rate at which FireWire packets happen
to be sent, but restrict them to the actual rate of a real MIDI port.
This is required by the specification, and prevents data loss when the
device's buffer overruns.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are several devices that expect to receive MIDI data only in the
first eight data blocks of a packet. If the driver restricts the data
rate to the allowed rate (as mandated by the specification, but not yet
implemented by this driver), this happens naturally. Therefore, there
is no reason to ever try to use more data packets with any device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the 't->length' is a big-endian value, it's used without any
conversion. This means that the driver always uses 'length' parameter.
Fixes: 555e8a8f7f14("ALSA: fireworks: Add command/response functionality into hwdep interface")
Reported-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code tends to use unsigned variables by default and it causes
signedness bugs when we use negative variables for error handling.
The "i" and "j" variables are used to iterated over small positive
values and so they should be type "int". The "len" variable doesn't
*need* to be signed but it should be signed to make the code easier to
read and audit.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code causes a static checker warning:
sound/firewire/oxfw/oxfw.c:46 detect_loud_models()
warn: signedness bug returning '(-2)'
The detect_loud_models() function should return false on falure, so that
we don't try to set up the loud code for hardware that doesn't support
it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds MIDI functionality with an assumption of 'if the device
has MIDI comformant data channels in its stream formation, the device has
one MIDI port'.
When no streams have already started, MIDI functionality starts stream
with current sampling rate.
When MIDI functionality has already starts some streams and PCM
functionality is going to start streams at different sampling rate,
this driver stops streams once and changes sampling rate, then restarts
streams for both PCM/MIDI substreams.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, a support for transmitted packets is added. This commit
add a support for capturing PCM samples.
When any streams are already started, this driver should not change sampling
rate of the device, thus this commit also adds a restriction of sampling rate
in this situation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices produced by Behringer/Mackie are based on OXFW970/971. This
commit adds support for them. Additionally, this commit changes the way to
name card with some information in config rom.
Ids of some Mackie(Loud) models are not identified, therefore this commit
applies name detection for these models.
The devices support capture/playback of PCM-samples and some of them
supports capture/playback of MIDI messages. These functionalities are
implemented by followed commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In past commit, this driver can keep stream formations for each sampling
rate. So its stream functionality can decide stream formations with given
some parameters.
This commit moves related codes from PCM functionality to stream
functionality. Furthermore, to set stream format correctly, this commit
uses AV/C Stream Format Information command instead of AV/C Input/Output
Plug Signal Format command.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc interface to get information about stream
formation. This commit also adds snd_oxfw_stream_get_current_formation()
to get current stream formation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
