When the system returns from suspend, it looses its configuration. Most
of it is restored by running a normal audio stream startup, but the DAI
format is left unset as that's configured on the audio device creation.
Hence, it suffices here to care for the registers which are touched by
davinci_mcasp_set_dai_fmt() and restore them when the system is resumed.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysclk rate does not change runtime so it should be initialized at
init time.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A relative calm release at this time with a flat diffstat.
The only significant change in the ALSA core side is the support for
more than 32 card instances, configurable via kconfig.
Other than that, in both ASoC and other parts, mostly some
improvements and fixes on the driver side.
- hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
- hda: Haswell HDMI audio fixes, runtime PM improvements
- hda: Intel BayTrail support, ALC5505 DSP support
- es1968: MediaForte M56VAP support
- usb-audio: Improved support for Yamaha/Roland devices
- usb-audio: M2Tech hiFace, Audio Advantage Micro II support
- hdspm: wordclock fixes
- ASoC: Pending fixes for WM8962
- ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
- ASoC: Generalisation of the Bluetooth and HDMI stub drivers
- ASoC: SSM2518 and RT5640 codec drivers.
- ASoC: Tegra CPUs with RT5640 machine driver
- ASoC: AC'97 refactoring bug fixes
- ASoC: ADAU1701 driver fixes
- Clean up of *_set_drvdata() in a wide range of drivers
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Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"A relative calm release at this time with a flat diffstat. The only
significant change in the ALSA core side is the support for more than
32 card instances, configurable via kconfig. Other than that, in both
ASoC and other parts, mostly some improvements and fixes on the driver
side.
- hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
- hda: Haswell HDMI audio fixes, runtime PM improvements
- hda: Intel BayTrail support, ALC5505 DSP support
- es1968: MediaForte M56VAP support
- usb-audio: Improved support for Yamaha/Roland devices
- usb-audio: M2Tech hiFace, Audio Advantage Micro II support
- hdspm: wordclock fixes
- ASoC: Pending fixes for WM8962
- ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
- ASoC: Generalisation of the Bluetooth and HDMI stub drivers
- ASoC: SSM2518 and RT5640 codec drivers.
- ASoC: Tegra CPUs with RT5640 machine driver
- ASoC: AC'97 refactoring bug fixes
- ASoC: ADAU1701 driver fixes
- Clean up of *_set_drvdata() in a wide range of drivers"
* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
ALSA: vmaster: Fix the regression of missing vmaster hook call
ALSA: hda - Add Dell SSID to support Headset Mic recording
ASoC: adau1701: remove control_data assignment
ASoC: adau1701: more direct regmap usage
ASoC: ac97: fixup multi-platform AC'97 module build failure
ASoC: pxa2xx: fixup multi-platform AC'97 build failures
ASoC: tegra20-ac97: Remove unused variable
ASoC: tegra20-ac97: Remove duplicate error message
ALSA: usb-audio: Add Audio Advantage Micro II
ASoC: tas5086: fix Mid-Z implementation
ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
ALSA: Replace the magic number 44 with const
ALSA: hda - Fix the max length of control name in generic parser
ALSA: hda - Guess what, it's two more Dell headset mic quirks
ALSA: hda - Yet another Dell headset mic quirk
ALSA: hda - Add support for ALC5505 DSP power-save mode
ASoC: mfld: Remove unused variable
ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
ALSA: usb-audio: claim autodetected PCM interfaces all at once
ALSA: usb-audio: remove superfluous Roland quirks
...
Move mach-davinci/dma.c to common/edma.c so it can be used
by OMAP (specifically AM33xx) as well.
Signed-off-by: Matt Porter <mporter@ti.com>
Acked-by: Chris Ball <cjb@laptop.org> # davinci_mmc.c
Acked-by: Mark Brown <broonie@linaro.org>
Acked-by: Olof Johansson <olof@lixom.net>
[nsekhar@ti.com: dropped davinci sffsdr changes]
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
sffsdr machine support does not build since at least v2.6.36
(~3 years). There is little hope of it being fixed, so remove
the support.
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
'mcasp_dt_ids' is always compiled in. Hence of_match_ptr is not
necessary.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
McASP serial audio engine needs different rotation values on TX and RX
channels. Commit dde109fb46 ("ASoC: McASP: Fix data rotation for
playback. Enables 24bit audio playback") changed the calculation to fix
the playback format, but broke the capture stream by doing it for both
TXFMT and RXFMT.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org [3.9 only]
The pcm3008_codec struct is not used outside of davinci-sffsdr.c, so make it
static.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few more bug fixes, the DAPM clock fix is actually a driver specific
one since currently there's only one user of the clock support due to
the problems relying on the clock API.
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Merge tag 'asoc-v3.10-4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A few more bug fixes, the DAPM clock fix is actually a driver specific
one since currently there's only one user of the clock support due to
the problems relying on the clock API.
According documentation bit ACLKRPOL is set to 0 (receiver samples data
on falling edge) and when set to 1 (receiver samples data on rising edge).
I2S data are always sampled on falling edge and valid during rising edge
of bit clock. So in case of capture data transmitter sample data on falling
edge and macsp must read then on rising edge.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When McASP is bit clock and frame clock master enable pin output for rx clocks.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length).
I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels.
Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback"
number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Channel size settings will be made at the end of
davinci_mcasp_hw_params() routine and thus overwrite frame
format settings made for DIT mode. This patch fixes this issue
by taking op_mode into account. Tested with official PSP 3.2
kernel and sii9022a HDMI transmitter.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AFSX won't be used in DIT mode. The related pins are AHCLKX and
the data pins.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Tested-by: Daniel Mack <zonque@gmail.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>
Davinci McASP has support for I2S multichannel playback.
For I2S playback/receive, each serializer is capable to play 2 channels
(L/R) audio data.Serializer function (Playback-receive-none) is configured
in DT, depending on hardware specification. It is possible to play less
channels than configured in DT. For that purpose,only specific number of
active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt
set to number of enabled serializers, otherwise no data are transfered to
McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)"
error. For TDM mode, McASP is capable to play or receive 32 channels for one
serializer. McAsp has support for max 16 serializer, therefore max channels
is 32 * 8.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
u32 rotate = (32 - word_length) / 4;
This implementation is wrong, but it works only for 16, or 32 bit audio data.
(rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in
4 bits (max value 0x7)and then masks them . That data are sended to i2s bus.
For 24 bit or 20 bit or other data formats, this code rotates data badly and
you hear somethink like noise. You need to use
u32 rotate = (word_length / 4) & 0x7;
to proper data rotation.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Depending on the Codec, the the BCLK/LRCLK ratio might not be freely
chosen by the CPU DAI.
For example, some Codec might want to be supplied with 32-bit samples
for both its channels regardless of the actual audio word size the CPU
sends. In such cases, the rest of the bits on the data lines must be
padded with zeros:
_______________________________
LRCLK / \
--' `---------- .....
BCLK ||||||||||||||||||||||||||||||||||||||||||||||| .....
DATA ____||||||||||||||||_________________|||||||||| .....
|<-- data -->|<-- pads --> |
This patch adds a new clock divider to configure the BCLK/LRCLK ratio.
If the machine code uses that divider, the driver uses the specified
value, instead of deriving that information from the audio word size.
Otherwise, the original behaviour is retained.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change davinci_config_channel_size() to derive the values for XSSZ and
XROT in DAVINCI_MCASP_[RT]XFMT_REG from the configured word length
rather than hard-coding them in a switch/case block.
Also, by directly passing the word length to
davinci_config_channel_size(), we can get rid of the
DAVINCI_AUDIO_WORD_* enum.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec_fmt and sample_rate variables are unused in both snd_platform_data
and davinci_audio_dev, so drop them.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removes the DaVinci private SRAM API and replaces it with
the genalloc API. The SRAM gen_pool is passed in pdata since
DaVinci is in the early stages of DT conversion.
[zonque@gmail.com: stub out gen_pool functions for
!CONFIG_GENERIC_ALLOCATOR]
Signed-off-by: Matt Porter <mporter@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Mike Looijmans <mike.looijmans@topic.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The FSDUR flag configures whether the frame clock uses a high phase of
only one bit or a full word. This has to be set depending on the DAI
format.
For other modes than DSP_B, the FSXDLY/FSRDLY fields have to be set to
1.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a .set_sysclk function to pass the direction of the clock down to
the driver. Only enable AHCLKX in the PDIR register when the CPU is
driving the clock.
This also removes the modification of the AHCLKXE/AHCLKRE bits in the
hw_params callback, and users must set the desired configuration using
snd_soc_dai_set_sysclk(), which this patch also does for the only user
in mainline (davinci-evm).
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the internal clock dividers of the McASP driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The OMAP2+ variant of McASP is different from Davinci variant w.r.to
some register offset.
Changes
- Add new MCASP_VERSION_3 to identify new variant. New DT compatible
"ti,omap2-mcasp-audio" to identify version 3 controller.
- The register offsets are handled depending on the version.
Note:
DMA parameters (dma fifo offset) are not updated and will be done later.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix typo caused by recent commit (cf53756 - ASoC: davinci: davinci-pcm
does not need to be a plaform_driver)
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add device tree probe for McASP driver.
Note:
DMA parameters are not populated from DT and will be done later.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Davinci McASP header & driver are shared by few OMAP platforms (like
TI81xx, AM335x). Splitting asp header into Davinci platform specific
header and Audio specific header helps to share them across platforms.
Audio specific defines is moved to to common
<linux/platform_data/davinci_asp.h> so that the header can be
accessed by all related platforms.
While here, correct the header usage (remove multiple header
re-definitions and unused headers) and remove platform names from
structures comments and enum. Also some some coding style errors.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Same as the commit 518de86 (ASoC: tegra: register 'platform' from DAIs,
get rid of pdev). It makes davinci-pcm not a platform_driver but helper
to register "platform", so that the platform_device for davinci-pcm can
be saved completely.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Add Runtime PM support to McASP host controller.
* Use Runtime PM API to enable/disable McASP clock.
This was tested on AM18x Board using suspend/resume
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Defines or parameters from <mach/mux.h> isn't used anywhere. Hence
remove the header include.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FIFO should be flushed before it is enabled for the first time.
This fixes the I/O errors reported by the ASoC core on a fresh boot
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a follow up on 53dea36c70 which fixes the other affected
pcm engines.
Description from 53dea36c70:
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Without this patch I was seeing null-pointer dereferenc in atmel-pcm.
Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
In this case, the original code did not contain a call to iounmap, nor does
one appear anywhere else in the file. I have assumed that it is safe to
use devm_ioremap for the allocation in any case.
Signed-off-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
In this case, the original code did not contain a call to iounmap, nor does
one appear anywhere else in the file. I have assumed that it is safe to
use devm_ioremap for the allocation in any case.
Signed-off-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses devm_kzalloc, devm_request_mem_region and
devm_ioremap for data that is allocated in the probe function of a platform
device and is only freed in the remove function.
Signed-off-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Missed .owner of struct snd_soc_card will prevent the module from being
removed from underneath its users.
Reported-by: Lothar Waßmann <LW@KARO-electronics.de>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.
The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
SND_DM365_EXTERNAL_CODEC does not exist, so it's a useless default.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The core will sync DAPM as part of the card initialization, there is no
need for machine drivers to do so during their setup.
OMAP drivers are omitted as I know Peter already has patches for them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link")
changed DAI format flag values and we cannot simply invert anymore e.g.
frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there
is no anymore fixed bit position for bit-clock or frame-sync inversion.
Fix this by relying only on DAI format flag values passed to us and by not
making any assumption on individual bit positions
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Vaibhav Bedia <vaibhav.bedia@ti.com>
Cc: Sekhar Nori <nsekhar@ti.com>
Cc: Kevin Hilman <khilman@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.
Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Although the McASP supports sign-extending samples in RX or TX [1]; the
davinci-mcasp driver does not touch the {R,X}PBIT or {R,X}PAD field of the
{R,X}FMT registers meaning that the McASP will serialize the bytes it is given
regardless of their signedness. So supporting unsigned formats is as simple
as adding them to the metadata of the davinci-mcasp driver.
Update the FMTBITs reported in the snd_soc_dai_driver and also update the case
statements in davinci-mcasp's hw_params() function so that the McASP can be
connected to CODECs that use unsigned values.
[1] http://www.ti.com/lit/ug/sprufm1/sprufm1.pdf
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
In the previous commit 'ASoC: davinci-pcm: convert to BATCH mode', the phase
offset of 2 was mentioned in the commit message but not well commented in the
source.
Add descriptive comments of the phase offset with and without ping-pong
buffers enabled.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The davinci-pcm driver's snd_pcm_ops pointer function currently calls into
the edma controller driver to read the current positions of the edma channels
to determine pos to return to the ALSA framework. In particular,
davinci_pcm_pointer() calls edma_get_position() and the latter has a comment
indicating that "Its channel should not be active when this is called" whereas
the channel is surely active when snd_pcm_ops.pointer is called.
The operation of davinci-pcm in capture and playback appears to follow close
the other pcm drivers who export SNDRV_PCM_INFO_BATCH except that davinci-pcm
does not report it's positions from pointer() using the last transferred
chunk. Instead it peeks directly into the edma controller to determine the
current position as discussed above.
Convert the davinci-pcm driver to BATCH mode: count the periods elapsed in the
prtd->period member and use its value to report the 'pos' to the alsa
framework in the davinci_pcm_pointer function.
There is a phase offset of 2 periods between the position used by dma setup
and the position reported in the pointer function. Either +2 in the dma
setup or -2 in the pointer function (with wrapping, both) accounts for this
offset -- I opted for the latter since it makes the first-time setup clearer.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extract functions that modify the prtd->period member in preparation for
conversion to BATCH mode playback.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The release of the dma channels was being performed in prepare and there was a
edma_resume call for the asp-channel only being executed on START, RESUME and
PAUSE_RELEASE.
The mcasp on da850evm with ping-pong buffers enabled was exhibiting an audible
glitch on every playback after the first. It was determined through trial and
error that the following two changes fix this problem:
1) Move the edma_start calls from prepare to trigger and 2) reverse the order
of starting the asp and ram channels.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the registration of davinci-mcasp.1 in the davinci-evm platform
setup for da830 and dm6467, davinci-pcm can handle more than the currently
reported maximum channels of 2.
Increase the maximum channels to 384 to match the maximum reported by
davinci-mcasp.1.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the data_type test in ping_pong_dma_setup, davinci-pcm is capable of
handling data of width up to and including 32bits.
"
if ((data_type == 0) || (data_type > 4)) {
printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
return -EINVAL;
}
"
Update the .format member of the snd_pcm_hardware instances it registers to
reflect this capability.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The setup of the pong channel uses EDMA_CHAN_SLOT instead of & 0x3f as the
setup of the ping channel does.
Make the setup of ping and pong symmetric. There is no functional change
introduced by this patch.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The davinci-i2s driver copies the platform data for playback and capture
sram sizes which is in turn used by davinci-pcm to allocate ping-pong
buffers.
Copy also the platform data in davinci-mcasp probe.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current davinci_mcasp_set_dai_fmt() sets bits ACLKX and ACLKR in the PDIR
register for the codec clock-master/frame-slave mode; however, this results in
the ACLKX and ACLKR pins being outputs according to SPRUFM1 [1] which
conflicts with "codec is clock master."
Similarly to the previous patch in this series, "fix _CBM_CFS hw_params" --
For codec clock-master/frame-slave mode (_CMB_CFS), clear bits ACLKX and ACLKR
in the PDIR register to set the pins as inputs and hence allow externally
sourced bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current davinci_mcasp_set_dai_fmt() sets bits ACLKXE and ACLKRE (CLKXM
and CLKRM as they are reffered to in SPRUFM1 [1]) for codec clock-slave/
frame-slave mode (_CBS_CFS) which selects internally generated bit-clock and
frame-sync signals; however, it does the same thing again for codec
clock-master/frame-slave mode (_CBM_CFS) in the very next case statement which
is incorrectly selecting internally generated bit-clocks in this mode.
For codec clock-master/frame-slave mode (_CBM_CFS), clear bits ACLKXE and
ACLKRE to select externally-generated bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current driver creates value for set/clr of PDIR using (x<<26) instead
of the #defines that are convieniently made available.
Update the driver to use the bitfield definitions of PDIR. There is no
functional change introduced by this patch.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current check for the number of tdm-slots specified by platform data is
always true (x >= 2 || x <= 32); therefore the else branch that warns of an
incorrect number of slots can never be taken.
Check that the number of tdm slots specified by platform data is between 2
and 32, inclusive.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use mfd_data for passing information from mfd drivers to soc
clients. The mfd_cell's driver_data field is being phased out.
Clients that were using driver_data now access .mfd_data
via mfd_get_data().
Signed-off-by: Andres Salomon <dilinger@queued.net>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
This patch modifies the Davinci i2s and mcasp drivers to make use of
ioremap() instead of IO_ADDRESS()
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case of any error in probe() function, clk_disable() and clk_put()
should be called if clk_enable() and clk_get() went through.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modifies the Davinci i2s and mcasp drivers
to make use of the resource_size() helper function for readability.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP1 is used on the DA830/OMAP-L137 platform for the codec.
This is different from the DA850/OMAP-L138 platform which uses McASP0.
This is fixed by adding a new snd_soc_dai_link struct.
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c
is not matching with the i2c ids in the board file. Without this fix the
soundcard does not get detected on da850/omap-l138/am18x evm.
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Tested-by: Dan Sharon <dansharon@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37)
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Multi-component commit f0fba2ad broke a few things which this patch should
fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be
good if those with access to other Davinci boards could test.
--
The multi-component commit put the initialisation of
snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the
McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c).
The initialisation had to be moved from the probe function in these drivers
because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver.
Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in
davinci-pcm.c) before hw_params is called. I have moved the initialisation to
a new snd_soc_dai_ops.startup function in each of these drivers. This fix
indicates that all platforms that use davinci-pcm must have been broken and
need to test with this fix.
--
The multi-component commit also changed the McBSP driver name from
"davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board
level references to the driver name. This change is understandable, as there
is a similarly named "davinci-mcasp" driver in davinci-mcasp.c.
There is probably no 'correct' name for this driver. The DM6446 datasheet
calls it the "ASP" and describes it as a "specialised McBSP". The DM355
datasheet calls it the "ASP" and describes it as a "specialised ASP". The
DM365 datasheet calls it the "McBSP". Rather than fix this problem by
reverting to "davinci-asp", I've elected to avoid future confusion with the
"davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also
consistent with the names of the functions in the driver. There are other
fixes required, so it was never going to be as simple as a revert anyway.
--
The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has
2 ports), so I've changed the the id of the platform_device from 0 to -1.
--
In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link
structure with the DM355 EVM as they use different cpu DAI names (the DM355
has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port).
This also means that the 2 boards need different snd_soc_card structures.
--
The codec_name entries in davinci-evm.c didn't match the i2c ids in the board
files. I have only checked and fixed the details of the names used for the
McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should
check the others.
Signed-off-by: Chris Paulson-Ellis <chris@edesix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In this code, 0 is returned on failure, even though other
failures return -ENOMEM or other similar values.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@a@
identifier alloc;
identifier ret;
constant C;
expression x;
@@
x = alloc(...);
if (x == NULL) { <+... \(ret = -C; \| return -C; \) ...+> }
@@
identifier f, a.alloc;
expression ret;
expression x,e1,e2,e3;
@@
ret = 0
... when != ret = e1
*x = alloc(...)
... when != ret = e2
if (x == NULL) { ... when != ret = e3
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since only 4 mainline ASoC codecs support the trigger
callback, we cannot rely upon them stopping the frame clock
if they are master and must assume it is running even if the
sound is paused. Thus we cannot start the ASP until the trigger
method.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Martin Ambrose <martin@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.
This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.
platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.
Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.
Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.
This patch has been tested on DM644x and OMAP-L138 EVMs.
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
i2s_accurate_sck switch can be used to have a better approximate
sampling frequency.
The clock is an externally visible bit clock and it is named
i2s continuous serial clock (I2S_SCK).
The trade off is between more accurate clock (fast clock)
and less accurate clock (slow clock).
The waveform will be not symmetric.
Probably it is possible to get a better algorithm for calculating
the divider, trying to keep a slower clock as possible.
This patch has been developed against the
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When McBSP peripheral gets the clock from an external pin,
there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
and MCBSP_CLKS.
evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
hardware connection and I use MCBSP_CLKS, so I have added
this possibility.
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm)
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added two clocking options for dm365 McBSP peripheral when used
with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
from external pin and generates frame sync).
A slave clock management can be important when the external codec needs
the system clock and the bit clock synchronized (tested with uda1345).
This patch has been developed against the:
http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
git tree and has been tested on bmx board (similar to dm365 evm, but using
uda1345 as external audio codec).
Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf
Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)
During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.
https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).
The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files. percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.
percpu.h -> slab.h dependency is about to be removed. Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability. As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.
http://userweb.kernel.org/~tj/misc/slabh-sweep.py
The script does the followings.
* Scan files for gfp and slab usages and update includes such that
only the necessary includes are there. ie. if only gfp is used,
gfp.h, if slab is used, slab.h.
* When the script inserts a new include, it looks at the include
blocks and try to put the new include such that its order conforms
to its surrounding. It's put in the include block which contains
core kernel includes, in the same order that the rest are ordered -
alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
doesn't seem to be any matching order.
* If the script can't find a place to put a new include (mostly
because the file doesn't have fitting include block), it prints out
an error message indicating which .h file needs to be added to the
file.
The conversion was done in the following steps.
1. The initial automatic conversion of all .c files updated slightly
over 4000 files, deleting around 700 includes and adding ~480 gfp.h
and ~3000 slab.h inclusions. The script emitted errors for ~400
files.
2. Each error was manually checked. Some didn't need the inclusion,
some needed manual addition while adding it to implementation .h or
embedding .c file was more appropriate for others. This step added
inclusions to around 150 files.
3. The script was run again and the output was compared to the edits
from #2 to make sure no file was left behind.
4. Several build tests were done and a couple of problems were fixed.
e.g. lib/decompress_*.c used malloc/free() wrappers around slab
APIs requiring slab.h to be added manually.
5. The script was run on all .h files but without automatically
editing them as sprinkling gfp.h and slab.h inclusions around .h
files could easily lead to inclusion dependency hell. Most gfp.h
inclusion directives were ignored as stuff from gfp.h was usually
wildly available and often used in preprocessor macros. Each
slab.h inclusion directive was examined and added manually as
necessary.
6. percpu.h was updated not to include slab.h.
7. Build test were done on the following configurations and failures
were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my
distributed build env didn't work with gcov compiles) and a few
more options had to be turned off depending on archs to make things
build (like ipr on powerpc/64 which failed due to missing writeq).
* x86 and x86_64 UP and SMP allmodconfig and a custom test config.
* powerpc and powerpc64 SMP allmodconfig
* sparc and sparc64 SMP allmodconfig
* ia64 SMP allmodconfig
* s390 SMP allmodconfig
* alpha SMP allmodconfig
* um on x86_64 SMP allmodconfig
8. percpu.h modifications were reverted so that it could be applied as
a separate patch and serve as bisection point.
Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.
Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec,
the idea is to have both enabled in the same kernel simultaneously. However,
the current soc-core doesn't support simultaneous codecs, once that
support will have added, a patch will be posted to enable both codecs in
the DM365 EVM.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the support for the interface needed by the DaVinci
Voice Codec CQ93VC.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On TI DM6467 EVM, S/PDIF DIT codec fails to open as it is unable to install
hardware params. This dummy codec has no set_fmt and set_sysclk implementations
and calls from the application to these functions cause errors. This patch adds
a new hardware params callback function for S/PDIF transciever codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes after a suspend-resume cycle, the ALSA application
restarts the stream when resume fails and McASP fails to work
as the clock is not enabled. This patch corrects this bug.
Testes on TI DA850/OMAP-L138 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add clock enable and disable calls to resume and suspend respectively.
Also add a member to the audio device data structure which tracks the clock
status.
Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which
add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied.
[1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/
2009-November/016958.html
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use edma_pause and edma_resume to make missing dma_events
less likely. This may not be needed, but it looks better.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix underruns by using dma to copy 1st to sram
in a ping/pong buffer style and then copying from
the sram to the ASP. This also has the advantage
of tolerating very long interrupt latency on dma
completion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename variable master_lch to asp_channel
Rename variable slave_lch to asp_link[0]
Rename local variables:
lch to link
count to asp_count
src to asp_src
dst to asp_dst
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the left and right 16 bit samples to be shifted out as 1
32 bit sample.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove requirement that dma_params is 1st in the structures
davinci_audio_dev and davinci_mcbsp_dev.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.
It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
* 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci: (62 commits)
DaVinci: DM646x - platform changes for vpif capture and display drivers
davinci: DM355 - platform changes for vpfe capture
davinci: DM644x platform changes for vpfe capture
davinci: audio: move tlv320aic33 i2c setup into board files
DaVinci: EDMA: Adding 2 new APIs for allocating/freeing PARAMs
DaVinci: DM365: Adding entries for DM365 IRQ's
DaVinci: DM355: Adding PINMUX entries for DM355 Display
davinci: Handle pinmux conflict between mmc/sd and nor flash
davinci: Add NOR flash support for da850/omap-l138
davinci: Add NAND flash support for DA850/OMAP-L138
davinci: Add MMC/SD support for da850/omap-l138
davinci: Add platform support for da850/omap-l138 GLCD
davinci: Macro to convert GPIO signal to GPIO pin number
davinci: Audio support for DA850/OMAP-L138 EVM
davinci: Audio support for DA830 EVM
davinci: Correct the number of GPIO pins for da850/omap-l138
davinci: Configure MDIO pins for EMAC
DaVinci: DM365: Add Support for new Revision of silicon
DaVinci: DM365: Fix Compilation issue due to PINMUX entry
DaVinci: EDMA: Updating default queue handling
...
McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.
Tested on DA830/OMAP-L137 EVM, DM6467 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables tlv320aic3101 support on DM365 EVM and
it was tested on DM365 EVM rev c.
Note: this patch was created based on temp/asoc branch.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, the codec setup data structure has to remain for successful
probe.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- restructure to support multiple channel controllers by using
additional struct resources for each CC
- interface changes visible to EDMA clients
Introduce macros to build IDs from controller and channel number,
and to extract them. Modify the edma_alloc_slot function to take an
extra argument for the controller.
Also update ASoC drivers to use API. ASoC changes
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Move queue related mappings to dm<soc>.c
EDMA in DM355 and DM644x has two transfer controllers while DM646x
has four transfer controllers. Moving the queue to tc mapping and
queue priority mapping to dm<soc>.c will be helpful to probe these
mappings from platform device so that the machine_is_* testing will
be avoided.
- add channel mapping logic
Channel mapping logic is introduced in dm646x EDMA. This implies
that there is no fixed association for a channel number to a
parameter entry number. In other words, using the DMA channel
mapping registers (DCHMAPn), a PaRAM entry can be mapped to any
channel. While in the case of dm644x and dm355 there is a fixed
mapping between the EDMA channel and Param entry number.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Reviewed-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf
This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma setup code assumes that the buffer size is a multiple
of the period size.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dai is a parameter to the functions, so use it instead of
looking it up.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clock name strings are no longer passed on platform_data. Instead,
we rely entirely on struct device and clkdev to find the right clock.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the dma link with correct data as soon as
the master channel has copied it. Otherwise, the
1st period will play twice.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the codec is master then prepare should call
mcbsp_start, not trigger.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Code previously just "ors" in this field without clearing
first. Fix, by never reading this register.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only start sample generator if needed, and more
cleanup on davinci_mcbsp_start.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move variable declaration closer to use.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add davinci_mcbsp_dev as argument to davinci_mcbsp_start
and davinci_mcbsp_stop.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add toggle_clock function to complete i2s reset earlier.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No functional changes. Rename variable w to something
more meaningful. Remove code obfuscating macro MOD_REG_BIT.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign proper errors when platform resource claims fail.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch does the following:
(1) Add support for the DM646x machine
(2) Modifications required to introduce the platform driver model to get
platform data for all the machines including dm355 and dm644x.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds driver support for the two instances of McASP on TI's DM646x.
The multichannel audio serial port (McASP) functions as a general-purpose audio
serial port optimized for the needs of multichannel audio application.
(http://www.ti.com/litv/pdf/spruer1b).
There are two instances of McASP on DM646x. The McASP0 module includes up to 4
serializers that can be individually enabled to either transmit or receive
in different modes. The McASP1 module is limited with only 1 pinned-out
serializer that can be enabled to only transmit in DIT mode (neither receiving
in any mode nor transmitting in either Burst or TDM mode is supported).
McASP0 consists of transmit and receive sections that may operate
synchronized, or completely independently with separate master clocks, bit
clocks, and frame syncs, and using different transmit modes with different
bit-stream formats.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce the platform driver model to get platform data for dm355 and dm644x.
Register platform driver and acquire the resources in the probe function Since
the platform specific code had been moved from machine driver to dm<soc>.c
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a build fix, resyncing the DaVinci EVM ASoC board code
with the version in the DaVinci tree. That resync includes
support for the DM355 EVM, although that board isn't yet in
mainline.
(NOTE: also includes a bugfix to the platform_add_resources
call, recently sent by Chaithrika U S <chaithrika@ti.com> but
not yet merged into the DaVinci tree.)
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This resyncs the DaVinci I2S code with the version in the DaVinci
tree. The behavioral change uses updated clock interfaces which
recently merged to mainline. Two other changes include adding a
comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in
order to support more boards than just the DM644x EVM.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a buildfix for the DaVinci PCM code, resyncing it with
the version in the DaVinci tree. The notable change is using
current EDMA interfaces, which recently merged to mainline.
(The older interfaces never made it into mainline.)
NOTE: open issue, the DMA should be to/from SRAM; see chip
errata for more info. The artifacts are extremely easy to
hear on DM355 hardware (not yet supported in mainline), but
don't seem as audible on DM6446 hardwaare (which does have
mainline support).
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix for the error when the audio module is unloaded. On unregistering
the platform_device, platform_device_release will free the platform
data.If platform data is static the kernel panics when it is freed.
Instead use the platform device helper function to add data.
This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM operations tables are not exported directly but are instead
included in the platform structure so should be declared static.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove dependency on sffsdr_fpga_set_codec_fs() when the
SFFSDR FPGA module is not selected.
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Let's have audio playback not sound like chipmunks, 'k? :)
ASP1 on the DM355 EVM uses a 27 MHz external audio clock, not
the slower clock used with ASP0 on the DM6446 EVM.
Also, that slower ASP0 clock on the DM6446 is 12.288 MHz,
not 22.5792 MHz ... 48 KHz sample rate (x256), not a double
speed 44.1 KHz sample rate (which could be done, but isn't
what the board init code now sets up).
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Minor bugfix: now that DaVinci kernels can support multiple
boards, board-specific ASoC components need to verify they're
running on the right board before initializing.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.
Add dsp_a, dsp_b mode options
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Minor, just move a block of code to make next patch clearer.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just at little cleanup of davinci_i2s_set_dai_fmt
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Document the current polarity choices.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add constants with a value of 0 to show more explicitly
what is being requested.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock
[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert the tlv320aic3x codec driver to the new (standard) device
driver binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Vladimir Barinov <vbarinov@ru.mvista.com>
Tested-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Two warning messages lack their trailing new-line, add it.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
There are 43 includes of asm/mach-types.h by files that don't
reference anything from that file. Remove these unnecessary
includes.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct within the machine drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
