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Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support.
According referance menual, "Limited support for the case when
output sampling rates is between 8kHz and 30kHz. The limitation
is the supported ratio (Fsin/Fsout) range as between 1/24 to 8."
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sometimes the audio play can not be resumed after it is
suspended. Add snd_soc_pm_ops to execute power management
operations, then this issue is fixed.
Signed-off-by: Songjun Wu <songjun.wu@atmel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
identifier fld;
@@
* x = devm_kzalloc(...);
... when != x == NULL
x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
identifier fld;
@@
* x = devm_kzalloc(...);
... when != x == NULL
x->fld
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add NULL test on call to devm_kzalloc.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x;
@@
* x = devm_kzalloc(...);
... when != x == NULL
*x
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return type "unsigned int" was used by the ssm2518_lookup_mcs()
function even though it will eventually return a negative error code.
Improve this implementation detail by deletion of the type modifier then.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Multi channel support on Renesas R-Car sound.
This patch is tested on Salvator-X board, but it can't use
Multi channel, because supported format is different between
codec chip and R-Car.
Thus, it was tested on board which doesn't mount codec chip,
with oscilloscope.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to the codec driver to handle mic level
detect related IRQs, and report these to user-space using a uevent
variable.
The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to
user-space, if the mic level detect feature is enabled, and the
audio captured at the chosen mic(s) is above a certain threshold.
User-space can then handle the event accordingly (e.g. process
audio capture stream).
This method was chosen over ALSA control notification for a couple
of reasons:
1) There's no requirement here for a control to read state from.
The event is the only thing that's required and of interest.
2) tinyalsa support for control notifications does not exist so on
platforms using this over alsa-lib there is a need to add code
to support this event handling.
Another possible option would be to use the standard Jack reporting
framework but this really does not fit for this kind of event.
Finally, use of the input device framework is not being encouraged,
due to difficulties in enabling apps to access input devices, so
this has also been avoided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An external amp (if any) is connected to the external outputs of the SoC
of course, rather then directly to the internal amp.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a device tree match table. This serves to make the driver's support
of device tree more explicit.
Signed-off-by: Caesar Wang <wxt@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AFE is actually allowed to be turn on before configuration of DAIs
since each DAI has its own enabling control. Turn on/off AFE in
runtime resume/suspend to avoid AFE being shut down when closing a DAI
while other DAIs are still active.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch, internal speaker and line-out work,
but front headphone output jack stays silent on the
Mac Pro 4,1.
This code path also gets executed on the MacPro 5,1 due
to identical codec SSID, but i don't know if it has any
positive or adverse effects there or not.
(v2) Implement feedback from Takashi Iwai: Reuse
alc889_fixup_mbp_vref and just add a new nid
0x19 for the MacPro 4,1.
Signed-off-by: Mario Kleiner <mario.kleiner.de@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend / resume cycle resets the settings of the FM tuner. Restore
frequency settings on resume.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In symmetry we save context first before suspend and restore it last after
resume.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of tuner only card there is no need to take care of the codec which is
anyway absent.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If user does not supply tea575x_tuner parameter the driver tries to detect the
tuner type. The failed codec initialization is considered as FM-only card
present, however the driver still registers an IRQ handler for it.
Move codec detection earlier to set tea575x_tuner parameter before check.
Here the following functions are introduced
reset_coded() resets AC97 codec
snd_fm801_chip_multichannel_init() initializes cards with multichannel support
Fixes: 5618955c4269 (ALSA: fm801: move to pcim_* and devm_* functions)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection)
brings autodetection to the driver. However the autodetection algorithm misses
the TUNER_ONLY bit if it is supplied by the user.
Thus, user gets weird messages and no card registered.
snd_fm801 0000:0d:01.0: detected TEA575x radio type SF64-PCR
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
...
snd_fm801 0000:0d:01.0: AC'97 0 does not respond - RESET
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 interface is busy (1)
snd_fm801 0000:0d:01.0: AC'97 0 access is not valid [0x0], removing mixer.
snd_fm801: probe of 0000:0d:01.0 failed with error -5
Do a copy of TUNER_ONLY bit to be applied after autodetection is done.
Fixes: d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Cc: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to store struct pci_dev in struct fm801. Generic struct device
can be easily translated to struct pci_dev whenever it's needed, in particular
for one user for now.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The compiler complains on unused condition as follows
sound/pci/fm801.c: In function ‘snd_fm801_interrupt’:
sound/pci/fm801.c:585:3: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
Put the curly braces around empty body as suggested.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch introduces two new helpers fm801_iowrite16() and fm801_ioread16() to
write and read the registers by offset. Previously similar was done to access
the hardware registers by their names.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we will have a warning on ->remove() since device is a PCI one.
WARNING: CPU: 4 PID: 1411 at /home/andy/prj/linux/fs/proc/generic.c:575 remove_proc_entry+0x137/0x160()
remove_proc_entry: removing non-empty directory 'irq/21', leaking at least 'snd_fm801'
Fixes: 5618955c4269 (ALSA: fm801: move to pcim_* and devm_* functions)
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an API consolidation only. The use of kmalloc + memset to 0
is equivalent to kzalloc.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The upstreamed code modified the control names from Mute to
Switch without changing the logic. To get audio working the Switch
needs to be off which isn't aligned with normal ALSA conventions.
Inverting the logic now so that Switch Off means mute and Switch On
means active audio using the specific volume setting.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the fields channels_min, channels_max, rate and formats are
irrelevant for compressed playback, they will depend on the
content. This was probably a copy-paste mistake to have
them in the first place
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add dai links to enable additional playback stream with deeper
buffer for lower power consumption.
The normal and DEEP_buffer streams are not mutually exclusive,
content will be mixed by the DSP.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add definitions for MERR_DPCM_DEEP_BUFFER AND PIPE_MEDIA3_IN
Add relevant cpu-dai and dai link names
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
All the functionality was merged in DPCM-based driver,
keep older driver to avoid breaking userspace but
tag it as unsupported/deprecated
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge DMI quirks for various machines such as Asus T100
and clean-up code
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
first renaming and reducing delta with byt-rt5640 code before
dmi-based quirks are enabled
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
initial cleanup to use same pins
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using the hw_fixup function in order to overwrite the default SSP
setting for Audio DSP port connected to the codec. Instead of
TDM 4ch use I2S 2ch 24 bits.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
