The mpc5200_dma overwrites the private_data field of the CODEC's AC'97
device with the DMA drivers private data, but never actually reads it again.
Given that the private_data field is supposed to be owned by the AC'97
driver, overwriting it may cause undefined behavior. This patch removes the
code that overwrites the field from the driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The mpc5200_psc_ac97 driver puts a snd_ac97 device on the stack in the
driver probe function, initializes the private data member of the device and
the never uses the device again. It should be safe to remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The of_node_put() calls in imx_es8328_probe() may take uninitialized
pointers when reached though the early error path. This patch adds
the proper NULL initialization for fixing these.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
An error code was forgotten to be passed in the error path of
imx_es8328_probe().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm8962 driver uses the input subsystem, but it is selected by
SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled,
resulting in this link error:
ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!
ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!
ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!
Do not force the selection of the codecs by SND_SOC_FSL_ASOC_CARD to avoid
such problem.
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to use 'i2s-slave' property, since master/slave configuration
are passed via machine layer.
This change does not break existing users because they do check for slave
mode inside sound/soc/fsl/mpc8610_hpcd.c/p1022_ds.c/p1022_rdk.c
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
code can raise a panic when the ssi_private->pdev is null
[...]
/*
* If codec-handle property is missing from SSI node, we assume
* that the machine driver uses new binding which does not require
* SSI driver to trigger machine driver's probe.
*/
if (!of_get_property(np, "codec-handle", NULL))
goto done;
[...]
ssi_private->pdev =
platform_device_register_data(&pdev->dev, name, 0, NULL, 0);
[...]
done:
if (ssi_private->dai_fmt)
_fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
Proposal was to not use ssi_private->pdev->dev here but adding a new parameter
of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and
cpu_dai->dev in fsl_ssi_set_dai_fmt().
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Reported-by: Jean-Michel Hautbois <jean-michel.hautbois@vodalys.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check if ipg clock is in clock-names property, then we can move the
ipg clock enable and disable operation to startup and shutdown, that
is only enable ipg clock when ssi is working and keep clock is disabled
when ssi is in idle.
But when the checking is failed, remain the clock control as before.
Tested-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 'big-endian-data' property is originally used to indicate whether the
LSB firstly or MSB firstly will be transmitted to the CODEC or received
from the CODEC, and there has nothing relation to the memory data.
Generally, if the audio data in big endian format, which will be using the
bytes reversion, Here this can only be used to bits reversion.
So using the 'lsb-first' instead of 'big-endian-data' can make the code
to be readable easier and more easy to understand what this property is
used to do.
This property used for configuring whether the LSB or the MSB is transmitted
first for the fifo data.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Build kernel with SND_SOC_FSL_ASOC_CARD=m && SND_SOC_FSL_{SSI,SAI,ESAI}=y
leads the following error:
sound/built-in.o: In function `fsl_sai_probe':
>> fsl_sai.c:(.text+0x5f662): undefined reference to `imx_pcm_dma_init'
sound/built-in.o: In function `fsl_esai_probe':
>> fsl_esai.c:(.text+0x6044b): undefined reference to `imx_pcm_dma_init'
The config SND_SOC_FSL_ASOC_CARD is for IMX SOC, So move it under condition
of 'if SND_IMX_SOC'.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The imx-es8328 driver fails to build on PPC because it explicitly depends on
SND_SOC_IMX_PCM_FIQ, which itself doesn't build on PPC. Instead, rely on
the SND_SOC_FSL_SSI config option to pull in the necessary libraries.
While we're at it, remove SND_SOC_FSL_UTILS, which also is not needed.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building kernel with SND_SOC_IMX_AUDMUX=n leads to the following error:
sound/built-in.o: In function `fsl_asoc_card_probe':
>> fsl-asoc-card.c:(.text+0x1467b5): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467d0): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467ed): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x146807): undefined reference to `imx_audmux_v2_configure_port'
Update Kconfig to select SND_SOC_IMX_AUDMUX when SND_SOC_FSL_ASOC_CARD=y.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This adds an initial machine driver for the ES8328 audio codec on Freescale
boards. The driver supports headphones and an audio regulator for an onboard
speaker amp.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The previous patch (ASoC: fsl_sai: Add asynchronous mode support) added
new Device Tree bindings for Asynchronous and Synchronous modes support.
However, these two shall not be present at the same time.
So this patch just simply makes them exclusive so as to avoid incorrect
Device Tree binding usage.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SAI supports these operation modes:
1) asynchronous mode
Both Tx and Rx are set to be asynchronous.
2) synchronous mode (Rx sync with Tx)
Tx is set to be asynchronous, Rx is set to be synchronous.
3) synchronous mode (Tx sync with Rx)
Rx is set to be asynchronous, Tx is set to be synchronous.
4) synchronous mode (Tx/Rx sync with another SAI's Tx)
5) synchronous mode (Tx/Rx sync with another SAI's Rx)
* 4) and 5) are beyond this patch because they are related with another SAI.
As the initial version of this SAI driver, it supported 2) as default while
the others were totally missing.
So this patch just adds supports for 1) and 3).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds software reset code in dai_probe() so as to make a true init
by clearing SAI's internal logic, including the bit clock generation, status
flags, and FIFO pointers.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that
can be used, ideally, for all Freescale CPU DAI drivers and external CODECs.
The idea of this generic sound card is a bit like ASoC Simple Card. However,
for Freescale SoCs (especially those released in recent years), most of them
have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
this is a specific feature that might be painstakingly controlled and merged
into the Simple Card driver.
So having this driver will allow all Freescale SoC users to benefit from the
simplification to support a new card and the capability of wide sample rates
support through ASRC.
The driver is initially designed for sound card using I2S or PCM DAI formats.
However, it's also possible to merge those non-I2S/PCM type sound cards, such
as S/PDIF audio and HDMI audio, into this card as long as the merge will not
break the original function and as long as there is something redundant that
can be abstracted along with I2S type sound cards.
As an initial version, it only supports three cards that I can test:
imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC
imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver
imx-audio-wm8962, just like the old imx-wm8962.c driver
The driver is also compatible with the old Device Tree bindings of WM8962 and
SGTL5000. So we may consider to remove those two drivers after this driver is
totally enabled. (It needs to be added into defconfig)
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit a603c8ee52.
fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask().
fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit
to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled
bit to 0.
For esai when the bit value is 1, the slot is enabled, when the bit value is 0,
the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will
work abnormally. So revert this patch, make the esai use default function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add SND_SOC_DAIFMT_CBM_CFS support for Freescale architecture.
Successfully tested on i.MX 6Quad Wandboard and UDOO boards connected to
the pcm1792a codec.
In CBM_CFS mode, when using a sample size of 16 bits, we cannot use
CCSR_SSI_SCR_I2S_MODE_MASTER since we get a frame sync every 16 bits.
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Fabio Falzoi <fabio.falzoi84@gmail.com>
Tested-by: Angelo Adamo <adamo.a60@gmail.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we pass the port number through file private data for debugfs we cast
it to and from a pointer so use uintptr_t in order to ensure that the
types are compatible, avoiding warnings on 64 bit platforms where pointers
are 64 bit and unsigned integers 32 bit.
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch 3117bb3109: "ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and
platform drivers" from Jul 29, 2014, leads to the following Smatch
complaint:
sound/soc/fsl/fsl_asrc_dma.c:304 fsl_asrc_dma_shutdown()
warn: variable dereferenced before check 'pair' (see line 302)
sound/soc/fsl/fsl_asrc_dma.c
301 struct fsl_asrc_pair *pair = runtime->private_data;
302 struct fsl_asrc *asrc_priv = pair->asrc_priv;
^^^^^^^^^^^^^^^
Dereference.
303
304 if (pair && asrc_priv->pair[pair->index] == pair)
^^^^
Check.
305 asrc_priv->pair[pair->index] = NULL;
306
So we just let the driver check pair before using it.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM needs extra dapm routes in the machine driver to route audio
between Front-End and Back-End. In order to differ the stream names
in the route map from CODECs, we here add specific stream names to
SSI driver so that we can implement ASRC via DPCM to it.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM needs extra dapm routes in the machine driver to route audio
between Front-End and Back-End. In order to differ the stream names
in the route map from CODECs, we here add specific stream names to
SPDIF driver so that we can implement ASRC via DPCM to it.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM needs extra dapm routes in the machine driver to route audio
between Front-End and Back-End. In order to differ the stream names
in the route map from CODECs, we here add specific stream names to
SAI driver so that we can implement ASRC via DPCM to it.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM needs extra dapm routes in the machine driver to route audio
between Front-End and Back-End. In order to differ the stream names
in the route map from CODECs, we here add specific stream names to
ESAI driver so that we can implement ASRC via DPCM to it.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a cut and paste bug so it returns success instead of the error
code.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building a kernel with SND_SOC_GENERIC_DMAENGINE_PCM=n leads to the following
error:
ERROR: "snd_dmaengine_pcm_prepare_slave_config" [sound/soc/fsl/snd-soc-fsl-asrc.ko] undefined!
Let SND_SOC_FSL_ASRC select SND_SOC_GENERIC_DMAENGINE_PCM in order to fix such
error.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
