"Playback Design" products need a 50ms delay after setting the USB
interface.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This field may use up to 32 bits, so it should be handled as unsigned
int.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.
Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.
The datainterval is also ignored but there are not currently any quirks
which choose to override this.
Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware also has a PCM capture device which is not implemented in
this patch.
It may be possible to generalise this to Saffire 6 USB support and some
of the other Focusrite interfaces, but as I don't have access to these
devices we should wait until capture support is working first.
Capture support is not implemented because the code assumes the endpoint
to have its own interface (instead, it shares the interface with playback)
and some thought will be needed to lift this limitation.
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new module_pcmcia_driver() macro to remove the boilerplate
module init/exit code in the pcmcia drivers.
Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
During the transition to the generic parser, the hook to the codec
specific automute function was forgotten. This resulted in the silent
output on some MacBooks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"chn" here is a number between 0 and 255, but ->chn_info[] only has
16 elements so there is a potential write beyond the end of the
array.
If the seq_mode isn't SEQ_2 then we let the individual drivers
(either opl3.c or midi_synth.c) handle it. Those functions all
do a bounds check on "chn" so I haven't changed anything here.
The opl3.c driver has up to 18 channels and not 16.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec. The dspload_is_loaded() function returns
true if the DSP transfer was never started. This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called. dsp_loaded should never be set to true in this case,
skip it. The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
*path is not yet initialized when we check if the widget is connected.
The compiler also warns about this:
sound/soc/soc-dapm.c: In function 'is_connected_output_ep':
sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using the regmap API directly, we can make use of the
.write_flag_mask for SPI, which allows us to drop the strange register
hacks that were necessary so far.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <alexander.sverdlin@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.
BugLink: https://bugs.launchpad.net/bugs/1155016
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the driver doesn't power down the widget at going down to D3
when the widget node has an EAPD capability and EAPD is actually set
on all codecs unless codec->power_filter is set explicitly.
This caused a problem on some Conexant codecs, leading to click
noises, and we set it as NULL there. But it is very unlikely that the
problem hits only these codecs.
Looking back at the development history, this workaround for EAPD was
introduced just for some laptops with STAC9200 codec, then we applied
it blindly for all. Now, since it's revealed to have an ill effect,
we should disable this workaround per default and apply only for the
known requiring systems.
The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(),
and this has to be set explicitly by the codec driver when needed.
As of now, only patch_stac9200() sets this one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the dynamic array allocations for pins, converters and PCM arrays
instead of the fixed size arrays. The modern HDMI codecs get more and
more pins, and we don't know the sensitive limit.
Most of the patch are spent for the straight conversions from the
fixed array access to snd_array helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Older .bin files report the global coefficients as absolute address writes
to zero; maintain compatibility with them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
"val" has to be signed for the error handling to work.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.
As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.
This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSI is using devm_request_irq() from
1ddd82868c
(ASoC: fsi: use devm_request_irq())
master->irq is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using devm_clk_get() we can save a call to clk_put().
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert all uses of devm_request_and_ioremap() to the newly introduced
devm_ioremap_resource() which provides more consistent error handling.
devm_ioremap_resource() provides its own error messages so all explicit
error messages can be removed from the failure code paths.
Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990@gmail.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The support for the Openmoko Neo1973 GTA01 got removed in commit
1ae5cbc52e ("ASoC: neo1973_wm8753:
remove references to the neo1973-gta01 machine"). Remove its last traces
in the Kconfig file too.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As pointed of by Vaibhav, commit 2952b27e2 ("ASoC: davinci-mcasp:
Add support for multichannel playback") duplicated the logic of
counting the active serializers. That can be avoided by shifting
the code around a bit.
Also, drop two unused defines introduced by the same commit.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove probe function from the init section.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
This device supports multiple rates as described in later AC97
standards. This patch allows playback of different sample frequencies
without conversion.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return an error from wm_hubs_read_dc_servo() if hubs->dcs_readback_mode is not
correctly initialized. You might as well bail out since nothing is likely to
work correctly afterwards.
sound/soc/codecs/wm_hubs.c:321:11: warning: 'reg_r' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm_hubs.c:251:13: note: 'reg_r' was declared here
sound/soc/codecs/wm_hubs.c:322:11: warning: 'reg_l' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm_hubs.c:251:6: note: 'reg_l' was declared here
gcc version 4.6.3
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On
amd64 the result of the ones complement operator is then truncated to
unsigned int according to the prototype of snd_soc_update_bits(). I think
gcc is correctly warning that the upper 32 bits are lost.
sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params':
sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow]
gcc version 4.6.3
Add 2 more BCLKDIV mask macros as explained by Lars:
The BCLKDIV has three fields. The bitclock divider (bit 0-1), the samplerate
(bit 2-4) and the source select (bit 5). Here we want to update the bitclock
divider field and the samplerate field. When I wrote the code I was lazy and
used ~ADAU1373_BCLKDIV_SOURCE as the mask, which for this register is
functionally equivalent to ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK.
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
'data' is malloced in snd_soc_bytes_put() and should be freed
before leaving from the error handling cases, otherwise it will cause
memory leak.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
'file' is malloced in wm_adsp_load_coeff() and should be freed
before leaving from the error handling cases, otherwise it will cause
memory leak.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current snd_soc_register_codec() adds codec to list, and calls
snd_soc_register_dais().
But, this listed codec should be removed if dais registration
was failed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the module_i2c_driver() macro to make the code smaller
and a bit simpler.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the connection list expansion in hda_codec.c and hda_proc.c, the
value returned from snd_hda_get_num_raw_conns() is used as the array
size to store the connection list. However, the function returns
simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the
widget list with ranges isn't considered there. Thus it may return a
smaller size than the actual list, which results in -ENOSPC in
snd_hda_get_raw_conections().
This patch fixes the bug by parsing the connection list correctly also
for snd_hda_get_num_raw_conns().
Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the
interface associations come out wrong, which results in the driver
erroring out with the message "Audio class v2 interfaces need an
interface association".
Work around this by searching for the interface association descriptor
also in some other place where it might have ended up.
Reported-and-tested-by: Dave Helstroom <helstroom@google.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference should be moved below the NULL test.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Expose the newly added TCO LTC and sync check functions to userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds new ALSA controls to query the LTC state from userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch prepares snd_hdspm_get_sync_check() to also check the TCO
sync state. The added feature will be exposed to the user in a later
commit.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considerably shorten the code by using a macro. Though this won't lower
the binary size, it makes the source more readable.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a
dedicated 96K mode. The RME cards default to 96K mode, but since not all
external MADI equipment supports this, provide a switch to users that
changes the on-wire protocol to SMUX.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using the additional Time Code Option module in slave mode or the
SYNC-In wordclock connector, the sample rate needs to be returned by
hdspm_external_sample_rate().
Since this sample rate may contain any value with 1Hz granularity, we
need to round it to a common rate as done by the OSX driver.
[Fixed missing function declarations by tiwai]
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
