This commit adds a new driver for devices based on Fireworks. This driver
just creates/removes card instance according to callbacks.
Fireworks is a board module which Echo Audio produced. This module
consists of three chipsets:
- Communication chipset for IEEE1394 PHY/Link and IEC 61883-1/6
- DSP or/and FPGA for signal processing
- Flash Memory to store firmwares
Current supported devices:
- Mackie Onyx 400F/1200F
- Echo AudioFire12/8(until 2009 July)
- Echo AudioFire2/4/Pre8/8(since 2009 July)
- Echo Fireworks 8/HDMI
- Gibson Robot Interface pack/GoldTop
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds three commands, which may be used by some firewire device
drivers. These commands are defined in 'AV/C Digital Interface Command Set
General Specification Version 4.2 (2004006, 1394TA)'.
1. PLUG INFO command (clause 10.1)
2. INPUT PLUG SIGNAL FORMAT command (clause 10.10)
3. OUTPUT PLUG SIGNAL FORMAT command (clause 10.11)
By the command 1, the drivers can get the number of plugs for AV/C unit or
subunit.
By the command 2 and 3, the drivers can get/set sampling frequency.
The 'firewire-speakers' already uses INPUT PLUG SIGNAL FORMAT command to set
sampling rate. So this commit also affects the driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices based on BeBoB use this type of AV/C transaction.
'Deferred Transaction' is defined in 'AV/C Digital Interface Command Set
General Specification' and is used by targets to make a response deferred
during processing it.
If a target may not be able to complete a command within 100msec since
receiving the command, then the target shall return INTERIM response,
to which final response will follow later. CONTROL/NOTIFY commands are
allowed for deferred transaction.
In the specification, devices allow to send INTERIM response just one time.
But this commit allows to handle several INTERIM response with two reasons.
One reason is to simplify codes, and another reason is to prepare for
devices which is out of specification.
There is an issue. In the specification, the interval between INTERIM
response and final response is 'Unspecified interval'. The specification
depends on each subunit specification for this interval.
But we promise to finish this function for caller. In this reason, I use
FCP_TIMEOUT_MS for this interval. Currently it's 125msec. When we find
devices which needs more time for this interval, then let us add some codes
to apply more interval for 'Unspecified interval'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plug Control Registers have two fields related to the number of established
connections, one is 'Broadcast connection counter' and another is
'Point-to-point connection counter'. The driver can know there are established
connections or not to check these fields.
This commit is for considering about JACK/FFADO streaming. Currently, when
JACK/FFADO starts its streaming to the device, cmp_connection_establish() is
failed expectedly. This seems to be enough but there are some devices which
needs to change sampling frequency before trying to establish connections.
For such devices, this functionality is needed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros, codes with condition of direction and new functions
to handle output connection. Once cmp_connection_init() is executed with its
direction, CMP input and output connection can be handled by the same way.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to 'cmp_connection' structure to indicate
the direction of connection. This patch also adds 'direction' argument to
cmp_connection_init() function to determine the direction.
The cmp_connection_init() function is exported and used in snd-firewire-speakers
so this patch also affect it.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Referring to IEC 61883-1, oMPR and iMPR, oPCR and iPCR have some fields with
the same role in the same position. This patch renames some macros, variables
and function arguments with "i" in its prefix to reuse them between oMPR and
iMPR, oPCR and iPCR.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, drivers can bring XRUN state for PCM substreams when error to
queue packets or detecting discontinuity of packet. The application may try to
recover this state by calling snd_pcm_prepare().
Depending on each driver, .prepare() includes restart streaming. Then there
is a state that PCM substreams are running but isochronous contexts are
stopped. In this case, when .pointer() is called, it refers to error pointer.
This commit is for a prevention of this bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds common PCM constraints according to current firewire-lib
implementation.
1.Maximum width for each sample is limited by 24.
AM824 in IEC 61883-6 can deliver 24bit data.
2. Minimum time for period is 5msec.
Apply the old value. For shorter latency, further works are needed.
3. In blocking mode, frames per period/buffer is aligned to 32.
Each packet can include some frames depending on its sampling rate. In
blocking mode, the number equals to SYT_INTERVAL. Currently firewire-lib
can schedule snd_pcm_period_elapsed() for each packet. So, for accurate
PCM interrupt, the number of frames per period/buffer should be aligned
to SYT_INTERVAL.
Currently firewire-lib is lack of better rules to achieve this. So LCM of
each value of SYT_INTERVALs (=32) is applied. This can be improved for
further work.
[Fixed the compile error due to the missing "&" by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, AMDTP functionality in firewire-lib supports mapping
for PCM data channels. With this mapping, firewire-lib can obsolete
a flag, CIP_HI_DUALWIRE, but Dice driver still keeps dual wire mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices arrange the position of PCM/MIDI data channel in AMDTP packet.
This commit allows drivers to set channel mapping.
To be simple, the mapping table is an array with fixed length. Then the number
of channels for PCM is restricted by 64 channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Generally, the devices can synchronize to handle 'presentation timestamp'
in CIP packets. This commit adds functionality to pick up this timestamp from
in-packets transmitted by the device, then use it for out packets.
In current implementation, this module generated the timestamp by itself. This
is 'SYT Match' mode. Then drivers with this module acts as synchronization
master. This commit allows this module to act as synchronization slave.
This commit restricts this mechanism is only available in blocking mode because
handling the timestamp in non-blocking mode is more complicated than in
blocking mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For duplex streams with synchronization, drivers should pick up
'presentation timestamp' from in-packets and use the timestamp for
out-packets. This commit is preparation for this.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.
And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.
There are differences between specifications about MIDI conformant data.
About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.
So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.
About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.
So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.
Currently, capturing 32bit samples are supported.
When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.
The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The if condition here was supposed to return on error but the return
statement is missing. The effect is that the ->mixername is set to
"???" instead of "DT019X".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.
To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.
Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Commit 10df350977 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:
baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware
Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA
postion buffer properly while DMA1 and DMA2 can work well. So this patch masks
the buggy DMA0 by keeping it as opened.
This is a tentative workaround, so keep the change small as Takashi suggested.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.
So this patch moves PCRC and PRRC settings to the end of hw_params().
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.
So this patch adds an extra check in the code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.
So this patch fixes the condition here and adds one line comments to
make the purpose here clear.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.
This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm().
Also for CODEC to CODEC links the widgets are connected cross-over via a DAI
link widget, meaning that the capture widget of one CODEC will be connected to
the playback widget of the other and vice versa. Whereas
snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of
the CPU DAI to the playback widget of the CODEC DAI and the capture widget of
the CPU DAI to the capture widget of the CODEC DAI. So not skipping
CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create
incorrect connections between the two CODECs which will cause DAPM to detect
active paths where there are none and unnecessarily power up widgets.
Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.")
Cc: <stable@vger.kernel.org> (for 3.14+)
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255,
SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292,
SID: 0x10280684), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
And on the machine with SID 0x10280684, and the Lineout and external
microphone should be routed to docking, this patch also fix this
problem.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Block offset calculations are done in the contiguous allocator so
are not required here.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch fixes the following dereference check ordering.
sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746)
git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
git remote update asoc
git checkout 0b708c87f6
vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c
a4b12990 Mark Brown 2014-03-12 740 };
a4b12990 Mark Brown 2014-03-12 741
a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform)
a4b12990 Mark Brown 2014-03-12 743 {
a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev);
a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data;
0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev;
0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0;
a4b12990 Mark Brown 2014-03-12 748
a4b12990 Mark Brown 2014-03-12 @749 if (!pdata)
a4b12990 Mark Brown 2014-03-12 750 return -ENODEV;
a4b12990 Mark Brown 2014-03-12 751
a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel
count") channel count is no longer being set if monitor_present is 0.
This is because setting the count was moved after the CA value is
determined, which is only after the monitor_present check in
hdmi_setup_audio_infoframe().
Unfortunately, in some cases, such as with a non-spec-compliant codec or
with a problematic video driver, monitor_present is always 0. As a
specific example, this seems to happen with gen1 ATV (SiI1390 codec),
causing left-channel-only stereo playback (multi-channel playback has
apparently never worked with this codec despite it reporting 8 channels,
reason unknown).
Simply setting converter channel count without setting the pin infoframe
and channel mapping as well does not theoretically make much sense as
this will just mean they are out-of-sync and multichannel playback will
have a wrong channel mapping.
However, adding back just setting the converter channel count even in
no-monitor case is the safest change which at least fixes the stereo
playback regression on SiI1390 codec. Do that.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Stephan Raue <stephan@openelec.tv>
Tested-by: Stephan Raue <stephan@openelec.tv>
Cc: <stable@vger.kernel.org> # 3.12+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Read the stream offset and presentation position from DSP memory rather
than using the old estimated position. This fixes timing issues with
pulseaudio.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
hw_params() can be called multiple times. Make sure we release the DSP
stream that was allocated on previous hw_params() calls before allocating
a new DSP stream.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Intel IOMMU requires that the ACPI device is used to allocate all
DMA memory buffers. This means we need to pass the DMA device pointer into child
component devices that allocate DMA memory.
We also only set the DMA mask for the ACPI device now instead of for each
component device.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix page table creation on Haswell and Broadwell to remove unsafe
virt_to_phys mappings and use more portable SG buffer. Use audio buffer
APIs to allocate DMA buffers.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we add the allocated blocks to the modules list of blocks.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we dont alloc blocks twice with requests spanning more
than one block.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.
Add a workaround to detect and fix the corruption.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls. For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.
Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance. In such a case, it's superfluous to save the mixer
values multiple times. This patch fixes it by checking the counter.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This (widely used) construction:
if(printk_ratelimit())
dev_dbg()
Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.
[ 533.803964] retire_playback_urb: 852 callbacks suppressed
[ 538.807930] retire_playback_urb: 852 callbacks suppressed
[ 543.811897] retire_playback_urb: 852 callbacks suppressed
[ 548.815745] retire_playback_urb: 852 callbacks suppressed
[ 553.819826] retire_playback_urb: 852 callbacks suppressed
So use dev_dbg_ratelimited() instead of this construction.
Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067e), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
