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The platform device should only be registered from OMAPDSS HDMI
driver. The platform driver registers and unregisters all ASoC
components needed for OMAP HDMI audio.
The hdmi audio driver implements cpu-dai component using the callbacks
provided by OMAPDSS and registers the component under DSS HDMI
device. Omap-pcm is registered for platform component also under DSS
HDMI device. Dummy codec is used as as codec component. The hdmi audio
driver implements also the card and registers it under its own
platform device.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
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While rewriting the code in the previous commit [316638a5030a: ALSA:
virmidi: fixed code style issues], the error check was wrongly
converted. This resulted an Oops.
Fixes: 316638a5030a ('ALSA: virmidi: fixed code style issues')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixed some minor code style issues and also removed some
assignments inside of if conditionals.
Signed-off-by: Kyle Chamberlin <kylechamberlin@project20million.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a help for works in followed patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a help for works in followed patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is a help for works in followed patches.
And this commit remove 'fw_unit_get()/fw_unit_put()' because these
are called by helper functions in 'snd-firewire-lib'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Followed commits add much codes. To make the work easy, this commit creates
own directory and move current file to it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit renames 'firewire-speakers' to 'oxfw' to enhance support for
devices which based on OXFW970/971. A line for MODULE_ALIAS is added.
Additionally, to help for works in followed paches, some members
in private structure are renamed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some developers test this driver, thus it's better to remove its
experimental state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a new functions and some arrangement for PCM restriction.
This arrangement is due to the number of channels which each Dice device has.
I note that minimum number for period becomes 2, instead of 1 because its PCM
functionality has SNDRV_PCM_INFO_BATCH, this means that the driver uses double
(or more) buffering so the minimum number for period should be 2.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a file and move some codes related to proc output.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a file and move some codes related to hwdep functionality.
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Additionally, this interface give a way to read Dice notification.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a file and move some codes related to PCM functionality.
Currently PCM playback is supported. PCM capture will be supported in followed
commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a file with some helper functions for streaming, and move some
codes into the file with some arrangements.
Well-known CMP is not used to start/stop streams for Dice chipset. It's
achieved by writing to specific address. We call this way as 'enable'.
When devices are 'enabled', streaming starts in registered isochronous channel.
Some helper functions are already implemented in previous commit.
Basically, the stream is compliant to IEC 61883-6, so-called as AMDTP. But Dice
has a specific quirk, so called-as 'Dual Wire'. This quirk is applied at
176.4/192.0kHz. In this mode, each packet includes double number of events than
number in the specification, and stream runs at a half of sampling rate.
There is another quirk at bus reset. Dice chipset handles drivers' request but
don't re-enable streaming. So stream should be stopped.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This commit adds a file with some helper functions for transaction, and move
some codes into the file with some arrangements.
For Dice chipset, well-known FCP or AV/C commands are not used to control
devices. It's achieved by read/write transactions into specific addresses.
Dice's address area is split into 5 areas. Each area has its own role. The
offset for each area can be got by reading head of the address area. By
reading these areas, drivers can get to know device status. By writing these
areas, drivers can change device status.
Dice has a specific mechanism called as 'notification'. When device status is
changed, Dice devices tells the event by sending transaction. This notification
is sent to an address which drivers register in advance. But this causes an
issue to drivers.
To handle the notification, drivers need to allocate its own callback function
to the address region in host controller. This region is exclusive. For the
other applications, drivers must give a mechanism to read the received
notification. For this purpose, Dice driver already implements hwdep interface.
Dice chipset doesn't allow drivers to register several addresses. In this
reason, when this driver is applied to a device, the other drivers should
_not_ try to register its own address to the device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In followed commits, dice driver is split into several files. For easily
managing these files, this commit adds subdirectory and move file into
the directory.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently, dice driver supports AMDTP out-stream. In followed commits, AMDTP
in-stream will be supported but current name of members in dice structure
are not propper. This commit renames these members to proper name.
Additionally, for easily distinguishing local symbols from structure tag,
rename dice tag into snd_dice.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The checkpatch.pl generates some warnings due to:
- C99 comment
- a line over 80 characters
- min() for parameters with different types
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/tiwai/sound
Pull sound fixes from Takashi Iwai:
"No excitement, here are only minor fixes: an endian fix for the new
DSD format we added in 3.18, a fix for HP mute LED, and a fix for
Native Instrument quirk"
* tag 'sound-3.18-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: pcm: Add big-endian DSD sample formats and fix XMOS DSD sample format
ALSA: hda - One more HP machine needs to change mute led quirk
ALSA: usb-audio: Use snd_usb_ctl_msg() for Native Instruments quirk
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The commit [7a2e9ddc: ALSA: usb-audio: Add native DSD support for
Denon/Marantz DACs] requires the new format definition that has
landed only in for-next branch.
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Denon/Marantz USB DACs need a specific vendor command to switch between PCM and
DSD mode. This patch adds a new quirk function to switch between the two modes
using the specific USB vendor command.
This patch applies to the following devices:
- Marantz SA-14S1
- Marantz HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds native DSD support for the following devices:
- Marantz SA-14S1
- Marants HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Move capture PCM pin from PCM4 to PCM0 for Broadwell/Haswell.
This will allow us to integrate with pulseaudio better for
usually default device is set to 0.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The xmax volume should be corrected to ARRAY_SIZE(volume_map)-1, otherwise,
the xmax value will be mapped to 0 wrongly.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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User can pass snoop option to enable/disable the snoop behavior, but
currently azx_check_snoop_available() always turns it off for some
devices. For better debuggability, change the parameter as bint, and
allow user to enable/disable forcibly the snoop when specified via the
module option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a new driver_caps bit, AZX_DCAPS_SNOOP_OFF, to set the snoop off
as default. This new bit is used for the checks in
azx_check_snoop_available(). Most of case-switches are replaced with
the new dcaps in each entry.
While working on it, for avoiding to spend more bits, combine three
bits AZX_DCAPS_SNOOP_SCH, AZX_DCAPS_SNOOP_ATI and
AZX_DCAPS_SNOOP_NVIDIA bits into a flat type of two bits. This
reduces the bits usages, and assign AZX_DCAPS_OFF to this empty bit
now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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According to the sgtl5000 datasheet the MCLK frequency range restriction of
8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode.
mxs-sgtl5000 machine sets the codec as slave, and mx28 generates MCLK in the
range of 256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK
input.
In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we
want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the
current check would return -EINVAL, which is not correct.
Remove the 8 to 27MHz frequency check, since this only applies to the
synchronous SYS_MCLK input case.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When trying to play a 8kHz file with codec in slave mode we get the following
error on a mx28evk:
$ aplay -Dhw:0,0 stereo_8k.wav
Playing WAVE 'stereo_8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo
[ 21.218647] sgtl5000 0-000a: PLL not supported in slave mode
[ 21.224559] sgtl5000 0-000a: 128 ratio is not supported. SYS_MCLK needs to be 256, 384 or 512 * fs
[ 21.233687] sgtl5000 0-000a: ASoC: can't set sgtl5000 hw params: -22
aplay: set_params:1123: Unable to install hw params:
This error happens because we are using 'sys_fs' instead of 'frame_rate' in the
valid ratio check.
Use the real'frame_rate' so that the ratio is correctly calculated and the
playback can run.
sgtl5000 codec manual states that in 'Synchronous SYS_MCLK input' mode that the
following SYS_CLK frequencies are allowed: 256*fs, 384*fs, 512*fs.
, where fs is the sampling frequency, which can be in the range of:
8, 11.025, 16, 22.5, 32, 44.1, 48, 96 kHz.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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According to the sgtl5000 datasheet the MCLK frequency range restriction of
8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode.
When running the codec as slave, the master should generate MCLK in the range of
256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK input mode.
In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we
want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the
current check would return -EINVAL, which is not correct.
Remove the 8 to 27MHz frequency check, since this only applies to the
synchronous SYS_MCLK input case.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge the proper mixer resume support for quirk codes.
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Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Remove the manual transition back to SND_SOC_BIAS_STANDBY in resume. This
is already be automatically handled by the ASoC core.
Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.
The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed. While we are at it also remove the unused codec field from
the cq93vc struct so the whole probe function can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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While two of the fields in the cq93vc driver state struct are initialized
none of them are ever acutally read again. So remove the whole struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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On Broadwell, the ADSP FW don't support loopback record volume tuning,
so here remove this control.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The "changed" variable should be set to false at the start.
Fixes: a35daac77a03 ('ASoC: sigmadsp: Add support for fw v2')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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We must ensure that the clocking configuration is valid as rapidly as possible.
And do software reset before the others registers updates, or the registers
will be reset to the default state.
Signed-off-by: Zidan Wang <b50113@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In olden times the snd_hda_param_read() function always set "*start_id"
but in 2007 we introduced a new return and it causes uninitialized data
bugs in a couple of the callers: print_codec_info() and
hdmi_parse_codec().
Fixes: e8a7f136f5ed ('[ALSA] hda-intel - Improve HD-audio codec probing robustness')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adding machine driver to instantiate I2S based realtek's ALC5631
sound card on Arndale board.
There are other variants of Audio Daughter Cards for Arndale
Board for which support already exists but there is no support for
Realtek's alc5631 codec hence support for ALC5631 based machine
driver is being added.
This patch also documents the device tree binding for the Arndale
board based machine driver.
Signed-off-by: Claude Youn <claude.youn@gmail.com>
Signed-off-by: Krishna Mohan Dani <krishna.md@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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If regmap is disabled there will be no users of the ASoC regmap helpers.
Furthermore regmap_exit() will no be defined causing the following compile
error:
sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap':
sound/soc/soc-core.c:2645:2: error: implicit declaration of function
'regmap_exit' [-Werror=implicit-function-declaration]
So disable the helpers if regmap is disabled.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Pass flags to gpiod_get() to automatically configure the GPIOs. This is shorter
and not passing any flags to gpiod_get() will eventually no longer be supported.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The device node pointer 'cpu' is not used in the function
asoc_simple_card_parse_daifmt().
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch exports a core function which handles the DT description
of multi-codec links (as: "sound-dai = <&hdmi 0>, <&spdif_codec>;")
and creates a CODEC component array in the DAI link.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
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