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for better PCM period granularity
These three commits were merged to improve PCM pointer granularity.
commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call")
commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position")
commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing")
The point of them is to handle queued packets not only in software IRQ
context of IR/IT contexts, but also in process context. As a result of
handling packets, period tasklet is scheduled when acrossing PCM period
boundary. This is to prevent recursive call of
'struct snd_pcm_ops.pointer()' in the same context.
When the pointer callback is executed in the process context, it's
better to avoid the second callback in the software IRQ context. The
software IRQ context runs immediately after scheduled in the process
context because few packets are queued yet.
For the aim, 'pointer_flush' is used, however it causes a race condition
between the process context and software IRQ context of IR/IT contexts.
Practically, this race is not so critical because it influences process
context to skip flushing queued packet and to get worse granularity of
PCM pointer. The race condition is quite rare but it should be improved
for stable service.
The similar effect can be achieved by using 'in_interrupt()' macro. This
commit obsoletes 'pointer_flush' with it.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a
chmap has been already assigned, it returns as an error due to the
conflicting snd_ctl_add() result. However, this also clears the
already assigned chmap_kctl field via pcm_chmap_ctl_private_free(),
and becomes inconsistent in the later operation.
This patch adds the check of the conflicting chmap kctl before
actually trying to allocate / assign. The check failure is treated as
a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is
basically a driver bug and having the stack trace would help
developers to figure out the bad code path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In former commit, drivers in ALSA firewire stack always starts IT context
before IR context. If IR context starts after packets are transmitted by
peer unit, packet discontinuity may be detected because the context starts
in the middle of packet streaming. This situation is rare because IT
context usually starts immediately. However, it's better to solve this
issue. This is suppressed with CIP_SKIP_INIT_DBC_CHECK flag.
This commit enables the same feature as CIP_SKIP_INIT_DBC_CHECK.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for outgoing packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for incoming packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In clause 6.3 of IEC 61883-6:2000, there's an explanation about processing
of presentation timestamp. In the clause, we can see "If a function block
receives a CIP, processes it and subsequently re-transmits it, then the
SYT of the outgoing CIP shall be the sum of the incoming SYT and the
processing delay." ALSA firewire stack has an implementation to partly
satisfy this specification. Developers assumed the stack to perform as an
Audio function block[1].
Following to the assumption, current implementation of ALSA firewire stack
use one software interrupt context to handle both of in/out packets. In
most case, this is processed in 1394 OHCI IR context independently of the
opposite context. Thus, this implementation uses longer CPU time in the
software interrupt context. This is not better for whole system.
Against the assumption, I confirmed that each ASIC for IEC 61883-1/6
doesn't necessarily expect it to the stack. Thus, current implementation
of ALSA firewire stack includes over-engineering.
This commit purges the implementation. As a result, packets of one
direction are handled in one software interrupt context and spends
minimum CPU time.
[1] [alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062660.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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packet parameter
In packet streaming protocol applied to TASCAM FireWire series, the value
of SYT field in CIP header is always zero, therefore it has no meaning.
There's no need to synchronize packets in both direction for the series.
In current implementation of ALSA firewire stack, driver for the series
uses incoming packet parameter for outgoing packet parameter to calculate
the number of data blocks. This can be simplified because the task of
corresponding driver is to transfer data blocks enough to sampling transfer
frequency.
This commit purges support of full duplex synchronization to prevent
over-engineering implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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parameter
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>
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parameter
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>
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Subwoofer does not work out of the box on ASUS N751/N551 laptops. This
patch fixes it. Patch tested on N751 laptop. N551 part is not tested,
but according to [1] and [2] this laptop requires similar changes, so I
included them in the patch.
1. https://github.com/honsiorovskyi/asus-n551-hda-fix
2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781
Signed-off-by: Yura Pakhuchiy <pakhuchiy@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HD-audio reconfig function got broken in the recent kernels,
typically resulting in a failure like:
snd_hda_intel 0000:00:1b.0: control 3:0:0:Playback Channel Map:0 is already present
This is because of the code restructuring to move the PCM and control
instantiation into the codec drive probe, by the commit [bcd96557bd0a:
ALSA: hda - Build PCMs and controls at codec driver probe]. Although
the commit above removed the calls of snd_hda_codec_build_pcms() and
*_build_controls() at the controller driver probe, the similar calls
in the reconfig were still left forgotten. This caused the
conflicting and duplicated PCMs and controls.
The fix is trivial: just remove these superfluous calls from
reconfig_codec().
Fixes: bcd96557bd0a ('ALSA: hda - Build PCMs and controls at codec driver probe')
Reported-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since the recent rewrite of HD-audio infrastructure,
CONFIG_SND_HDA_RECONFIG has a slightly different meaning. In the
earlier versions, it implicitly assumed only the usage via hwdep sysfs
entries. Meanwhile, in the recent version, this option is meant to
enable the reconfig code in HD-audio driver, which may be used by the
patch loader without hwdep interface.
This patch tries to clarify the usage pattern a bit better, hopefully
avoid the further confusion.
Reported-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A switch statement looks a bit cleaner than an if statement
spread over 3 lines, as such update this to a switch.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We can't return a negative error code from the poll callback the return
type is unsigned and is checked against the poll specific flags we need
to return POLLERR if we encounter an error.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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stream can't be NULL here as we have just taken the address of it, so no
need for the check.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have a function that returns the appropriate flags for the stream
direction, so we should use it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We can't return a negative error code from the poll callback the return
type is unsigned and is checked against the poll specific flags we need
to return POLLERR if we encounter an error.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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context
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>
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context
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>
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For reducing the noise from the headset output on ASUS UX501VW,
call the existing fixup, alc_fixup_headset_mode_alc668(), additionally.
Thread: https://bbs.archlinux.org/viewtopic.php?id=209554
Signed-off-by: Kaho Ng <ngkaho1234@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If the DMAs are not being quiesced properly, it may lead to
stability issues, so the recommendation is to wait till DMAs are
stopped.
After setting the stop bit of RIRB/CORB DMA, we should wait for
stop bit to be set.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ebus is a member of extended device and was never initialized, so
do this at device creation.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Allow for SS+ USB devices
Signed-off-by: Oliver Neukum <oneukum@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Allow handling SS+ USB devices correctly.
Signed-off-by: Oliver Neukum <oneukum@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Support new codecs for ALC234/ALC274/ALC294.
This three codecs was the same IC.
But bonding is not the same.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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"header->number" can be up to USHRT_MAX and it comes from the ioctl so
it needs to be capped.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The stack object “r1” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The stack object “r1” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The stack object “tread” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Noticed two typos in Documentation, so fix them up
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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M-Audio Profire 610 has an unexpected value in version field of its config
ROM, thus ALSA dice driver is not assigned to the model due to a mismatch
of modalias.
This commit adds an entry to support the model. I expect the entry is
also for Profire 2626.
I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a
part of Extended Application Protocol (EAP).
$ cd linux-firewire-utils/src
$ ./crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
------------------------------------------------------------
400 04047689 bus_info_length 4, crc_length 4, crc 30345
404 31333934 bus_name "1394"
408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255,
max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400)
40c 000d6c04 company_id 000d6c |
410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002
root directory
------------------------------------------------------------
414 000695fe directory_length 6, crc 38398
418 03000d6c vendor
41c 8100000a --> descriptor leaf at 444
420 17000011 model
424 8100000d --> descriptor leaf at 458
428 0c0087c0 node capabilities per IEEE 1394
42c d1000001 --> unit directory at 430
unit directory at 430
------------------------------------------------------------
430 0004fb14 directory_length 4, crc 64276
434 12000d6c specifier id
438 130100d1 version
43c 17000011 model
440 8100000c --> descriptor leaf at 470
descriptor leaf at 444
------------------------------------------------------------
444 0004b8e4 leaf_length 4, crc 47332
448 00000000 textual descriptor
44c 00000000 minimal ASCII
450 4d2d4175 "M-Au"
454 64696f00 "dio"
descriptor leaf at 458
------------------------------------------------------------
458 00053128 leaf_length 5, crc 12584
45c 00000000 textual descriptor
460 00000000 minimal ASCII
464 50726f46 "ProF"
468 69726520 "ire "
46c 36313000 "610"
descriptor leaf at 470
------------------------------------------------------------
470 00053128 leaf_length 5, crc 12584
474 00000000 textual descriptor
478 00000000 minimal ASCII
47c 50726f46 "ProF"
480 69726520 "ire "
484 36313000 "610"
$ cat /proc/asound/card1/dice
sections:
global: offset 10, size 90
tx: offset 100, size 142
rx: offset 242, size 282
ext_sync: offset 524, size 4
unused2: offset 0, size 0
global:
owner: ffc0:000100000000
notification: 00000040
nick name: FW610
clock select: internal 48000
enable: 1
status: locked 48000
ext status: 00000040
sample rate: 48000
version: 1.0.4.0
clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal
clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\
...
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are many USB audio devices with buggy firmware that don't react
with the sample rate reading properly. This often results in the
flood of error messages and slowing down the operation.
The sample rate read back is basically only for confirming the sample
rate setup, and it's not critically important. As a compromise, in
this patch, we stop the sample rate read back once when the device
gives errors more than tolerance (twice, as of now). This should
improve most of error cases while we still can catch the firmware
bugginess.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.6
This is a fairly large collection of fixes but almost all driver
specific ones, especially to the new Intel drivers which have had a lot
of recent development. The one core fix is a change to the debugfs code
to avoid crashes in some relatively unusual configurations.
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asoc-linus
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'asoc/fix/hdac', 'asoc/fix/nau8825' and 'asoc/fix/rt5616' into asoc-linus
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For taking back the recent change of HDA HDMI fixes for i915 HSW/BDW.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent bug report suggests that BCLK setup for i915 HSW/BDW needs
to be updated at each HDMI hotplug, not only at initialization and
resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at
ELD notification, too. Otherwise the HDMI audio may be out of sync
and played in a wrong pitch.
However, the HDA codec driver has no access to the controller
registers, and currently the code managing these registers is in
hda_intel.c, i.e. local to the controller driver. For allowing the
explicit BCLK update from the codec driver, as in this patch, the
former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and
exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA
controller driver and intel_pin_eld_notify() in HDMI codec driver.
Along with this change, snd_hdac_get_display_clk() gets dropped as
it's no longer used.
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410
Cc: <stable@vger.kernel.org> # v4.5+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixes audio output on a ThinkPad X260, when using Lenovo CES 2013
docking station series (basic, pro, ultra).
Signed-off-by: Conrad Kostecki <ck+linuxkernel@bl4ckb0x.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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au88x0 hardware seems returning the current pointer at the buffer
boundary instead of going back to zero. This results in spewing
warnings from PCM core.
This patch corrects the return value from the pointer callback within
the proper value range, just returning zero if the position is equal
or above the buffer size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch tries to address the still remaining issues in ALSA hrtimer
driver:
- Spurious use-after-free was detected in hrtimer callback
- Incorrect rescheduling due to delayed start
- WARN_ON() is triggered in hrtimer_forward() invoked in hrtimer
callback
The first issue happens only when the new timer is scheduled even
while hrtimer is being closed. It's related with the second and third
items; since ALSA timer core invokes hw.start callback during hrtimer
interrupt, this may result in the explicit call of hrtimer_start().
Also, the similar problem is seen for the stop; ALSA timer core
invokes hw.stop callback even in the hrtimer handler, too. Since we
must not call the synced hrtimer_cancel() in such a context, it's just
a hrtimer_try_to_cancel() call that doesn't properly work.
Another culprit of the second and third items is the call of
hrtimer_forward_now() before snd_timer_interrupt(). The timer->stick
value may change during snd_timer_interrupt() call, but this
possibility is ignored completely.
For covering these subtle and messy issues, the following changes have
been done in this patch:
- A new flag, in_callback, is introduced in the private data to
indicate that the hrtimer handler is being processed.
- Both start and stop callbacks skip when called from (during)
in_callback flag.
- The hrtimer handler returns properly HRTIMER_RESTART and NORESTART
depending on the running state now.
- The hrtimer handler reprograms the expiry properly after
snd_timer_interrupt() call, instead of before.
- The close callback clears running flag and sets in_callback flag
to block any further start/stop calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Eric Engestrom <eric@engestrom.ch>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are no users of rtctimer left. Remove its code as this is the
in-kernel user of the legacy PC RTC driver that will hopefully be removed
at some point.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/evalenti/linux-soc-thermal
Pull thermal fixes from Eduardo Valentin:
"Specifics in this pull request:
- Fixes in mediatek and OF thermal drivers
- Fixes in power_allocator governor
- More fixes of unsigned to int type change in thermal_core.c.
These change have been CI tested using KernelCI bot. \o/"
* 'fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/evalenti/linux-soc-thermal:
thermal: fix Mediatek thermal controller build
thermal: consistently use int for trip temp
thermal: fix mtk_thermal build dependency
thermal: minor mtk_thermal.c cleanups
thermal: power_allocator: req_range multiplication should be a 64 bit type
thermal: of: add __init attribute
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