| Age | Commit message (Collapse) | Author |
|---|
|
The default state when serializers are in inactive slots is Hi-Z.
In some cases, there are no additional components driving the data
lines to a safe state so they might have noise.
While in inactive slots, the McASP AXR pins configured as outputs
can be driven low through the serializer pin drive mode setting
(DISMOD) to prevent such noise.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
An underrun (playback) event occurs when the serializer transfer
data from the XRBUF buffer to the XRSR shift register, but the
XRBUF hasn't been filled. Similarly, the overrun (capture) event
occurs when data from the XRSR shift register is transferred to
the XRBUF but it hasn't been read yet.
These events are handled as XRUN events that cause the pcm to stop.
The stream has to be explicitly restarted by the userspace which
ensures that after stopping/starting McASP the data transfer is
aligned with DMA. The other possibility was to internally stop and
start McASP without DMA even knowing about it.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Follow the sequence described in the TRMs when starting TX. This sequence
will make sure that we are not facing with initial channel swap caused by
no data available in McASP for transmit.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We can have more linear code flow by using variables in
mcasp_common_hw_param() related to the AFIFO configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
The FIFO registers base address is different in dm646x compared to newer
SoCs with McASP IP. Instead of using two paths (switch/case) to handle the
difference we can simply pick the correct base address beforehand and use
offsets to address the register we need to configure.
With this change the indentation depth can be reduced as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
Since it is a private struct strictly used by the davinci-mcasp driver it
can be moved from header file to the source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
It is better for readability to have the register definitions out from the
source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
These are not used, probably leftovers from the past.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
It is not used outside of the .c file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
|
|
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>
|
|
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>
|
|
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>
|
|
Signed-off-by: Daniel Mack <zonque@gmail.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>
|
|
* 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>
|
|
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>
|
|
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>
|
|
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>
|
|
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>
|
|
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>
|
|
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>
|
