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When build fsl-asoc-card as module, there is following error:
sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe':
>> sound/soc/fsl/fsl-asoc-card.c:547:13: warning: 'asrc_np' may be used uninitialized in this function [-Wmaybe-uninitialized]
of_node_put(asrc_np);
^
vim +/asrc_np +547 sound/soc/fsl/fsl-asoc-card.c
531 if (width == 24)
532 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
533 else
534 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
535 }
536
537 /* Finish card registering */
538 platform_set_drvdata(pdev, priv);
539 snd_soc_card_set_drvdata(&priv->card, priv);
540
541 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
542 if (ret)
543 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
544
545 fail:
546 of_node_put(codec_np);
> 547 of_node_put(asrc_np);
548 of_node_put(cpu_np);
549
550 return ret;
551 }
552
553 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
554 { .compatible = "fsl,imx-audio-cs42888", },
555 { .compatible = "fsl,imx-audio-sgtl5000", },
Add 'asrc_fail' branch for error jump after asrc_np initialized.
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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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This reverts commit a603c8ee526f5ea9ad9b40710308766299ad8a69.
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>
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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>
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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>
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sound/soc/fsl/fsl_asrc.c:250 fsl_asrc_config_pair()
warn: variable dereferenced before check 'config' (see line 243)
git remote add next git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git
git remote update next
git checkout 3117bb3109dc223e186302f5dc8ce9ed04adca90
vim +/config +250 sound/soc/fsl/fsl_asrc.c
237 */
238 static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
239 {
240 struct asrc_config *config = pair->config;
241 struct fsl_asrc *asrc_priv = pair->asrc_priv;
242 enum asrc_pair_index index = pair->index;
@243 u32 inrate = config->input_sample_rate, indiv;
244 u32 outrate = config->output_sample_rate, outdiv;
245 bool ideal = config->inclk == INCLK_NONE;
246 u32 clk_index[2], div[2];
247 int in, out, channels;
248 struct clk *clk;
249
@250 if (!config) {
251 pair_err("invalid pair config\n");
252 return -EINVAL;
253 }
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The patch 3117bb3109dc: "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>
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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>
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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>
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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>
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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>
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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>
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reproduce: make C=1 CF=-D__CHECK_ENDIAN__
sparse warnings: (new ones prefixed by >>)
>> sound/soc/fsl/fsl_asrc.c:563:28: sparse: restricted snd_pcm_format_t degrades to integer
>> sound/soc/fsl/fsl_asrc.c:570:28: sparse: restricted snd_pcm_format_t degrades to integer
vim +563 sound/soc/fsl/fsl_asrc.c
557 .probe = fsl_asrc_dai_probe,
558 .playback = {
559 .stream_name = "ASRC-Playback",
560 .channels_min = 1,
561 .channels_max = 10,
562 .rates = FSL_ASRC_RATES,
> 563 .formats = FSL_ASRC_FORMATS,
564 },
565 .capture = {
566 .stream_name = "ASRC-Capture",
567 .channels_min = 1,
568 .channels_max = 10,
569 .rates = FSL_ASRC_RATES,
> 570 .formats = FSL_ASRC_FORMATS,
571 },
572 .ops = &fsl_asrc_dai_ops,
573 };
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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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>
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Fix the following build errors that were observed by building with
make ARCH=microblaze allyesconfig:
>> sound/soc/fsl/fsl_asrc.c:906:5: warning: "CONFIG_PM_RUNTIME" is not defined [-Wundef]
#if CONFIG_PM_RUNTIME
^
>> sound/soc/fsl/fsl_asrc.c:934:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef]
#if CONFIG_PM_SLEEP
^
>> sound/soc/fsl/fsl_asrc.c:906:5: warning: "CONFIG_PM_RUNTIME" is not defined [-Wundef]
#if CONFIG_PM_RUNTIME
^
>> sound/soc/fsl/fsl_asrc.c:934:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef]
#if CONFIG_PM_SLEEP
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>
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The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a
signal associated with an input clock into a signal associated with a different
output clock. The driver currently works as a Front End of DPCM with other Back
Ends DAI links such as ESAI<->CS42888 and SSI<->WM8962 and SAI. It converts the
original sample rate to a common rate supported by Back Ends for playback while
converts the common rate of Back Ends to a desired rate for capture. It has 3
pairs to support three different substreams within totally 10 channels.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Varka Bhadram <varkabhadram@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The previous enable flow:
1, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO)
2, Mask IRQ of Tx/Rx to enable its interrupt.
3, Enable DMA request of Tx/Rx.
As this flow would enable DMA request later than TERE, the Tx FIFO
would be easily emptied into underrun while Rx FIFO would be easily
stuffed into overrun due to the delayed DMA transfering.
This issue happened merely occational before the patch 'ASoC: fsl_sai:
Reset FIFOs after disabling TE/RE' because there were useless data
remaining in the FIFO for the gap. However, it manifested after FIFO
reset's implemented.
After this patch, the new flow:
1, Enable DMA request of Tx/Rx.
2, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO)
3, Mask IRQ of Tx/Rx to enable its interrupt.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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TE/RE bit of T/RCSR will remain set untill the current frame is physically
finished. The FIFO reset operation should wait this bit's totally cleared
rather than ignoring its status which might cause TE/RE disabling failed.
This patch adds delay and timeout to wait for its completion before FIFO
reset.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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For trigger start, we don't need to check if it's the first time to
enable TE/RE or second time. It doesn't hurt to enable them any way,
which in the meantime can reduce race condition for TE/RE enabling.
For trigger stop, we will definitely clear FRDE of current direction.
Thus the driver only needs to read the opposite one's.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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ASoC: Fixes for v3.16
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
# gpg: Signature made Thu 19 Jun 2014 11:57:31 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
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In the rx irq handling part, we should clear the flags in RCSR not TCSR.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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SAI will not clear their FIFOs after disabling TE/RE. Therfore, the driver
should take care the task so as not to let useless data remain in the FIFO.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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I received a report this morning from one of the Novena developers that
the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was
sub-optimal under high system load.
While there are issues relating to system load remaining, upon reviewing
the ASoC imx-pcm-dma.c driver, it was noticed that it not using the
residue support, because SDMA doesn't support it. This has the effect
that SDMA has to make multiple calls into the ASoC and ALSA code, one
for each period.
Since ALSA's snd_pcm_elapsed() does not need to be called multiple times
and it is entirely sufficient to call it once to update ALSA with the
current buffer position via the pointer method, we can do better here.
We can also avoid stopping the DMA entirely, just like real cyclic DMA
implementations behave. While this means that we replay some old samples,
this is a nicer behaviour than having audio stop and restart.
The changes to achieve this are relatively minor - imx-sdma.c can track
where the DMA is to the nearest descriptor boundary - it does this
already when deciding how many callbacks to issue. In doing this,
buf_tail always points at the descriptor which will complete next.
The residue is defined by the bytes remaining to the end of the buffer,
when the buffer is viewed as a single block of memory [start...end].
So, when we start out, there's a full buffer worth of residue, and this
counts down as we approach the end of the buffer, eventually becoming
zero at the end, before returning to the full buffer worth when we
wrap back to the start.
Moving the walking of the descriptors into the interrupt handler means
that we can update the BD_DONE flag at interrupt time, thus avoiding
a delayed tasklet stopping the cyclic DMA.
This means that the residue can be calculated from (total descriptors -
buf_tail) * descriptor size. This is what the change below does. We
update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide
the residue.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
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Commit 31ee2bfd724ab ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA
where needed") started selecting SND_SOC_IMX_PCM_DMA and
SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX.
This has turned out too aggressive, as FIQ is only available
for i.mx2 through i.mx5, but not i.mx6 or vybrid.
Further, two more drivers have become user-selectable in the
meantime, and they both depend on DMA for the imx platform
as well.
This changes the selection of FIQ to depend on the TZIC or
AVIC interrupt controllers that actually export the imx
specific FIQ interfaces, and adds the missing select statements
for SAI and ESAI.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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do_div() requires that the first parameter is a 64-bit integer,
which but clkrate was defined as an unsigned long. This caused
the following warnings:
CC sound/soc/fsl/fsl_ssi.o
sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk':
sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast
sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type
sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type
include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *'
Signed-off-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Add support for the output sample rates 96kHz and 192kHz.
Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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1) Apply better indentations
2) Drop braces for single statement.
3) Use simpler ternary to reduce code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi.
Unfortunately, the structure is still used. This causes
mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with
sound/soc/fsl/fsl_dma.c:926:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
ound/soc/fsl/fsl_dma.c:927:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
Fix by using constants, similar to original commit.
Cc: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The calculation code does
u64 = (u32 - u32) * 100000;
The 64 bits are of no help here as the type is casted only after the
multiplication, and therefore the result may overflow, possibly causing
inoptimal or wrong clock setup in an unfortunate case (the maximum
result value of the first substraction is currently 47999).
Fix the code to cast before multiplication.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We should not copy the return value into this val since it's supposed to
get the value of the register not the success result of regmap_read().
Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Conflicts:
sound/soc/fsl/Kconfig
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Eukrea-i.MX51 board was converted to use DT, ie we no longer have a
MACH_EUKREA_MBIMXSD51_BASEBOARD symbol.
Transformation of other boards planned for the near future, so this
patch removes all these dependencies and restricts build of this
driver to ARCH_MXC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This patch replaces the ssi specific functions write_ssi, read_ssi and
write_ssi_mask by standard regmap function calls.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Reorder all variables in struct fsl_ssi_private to have groups that make
sense together. The patch also updates the struct documentation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The baudclock may be used and set by different streams.
Allow only the first stream to set the bitclock rate. Other streams have
to try to get to the correct rate without modifying the bitclock rate
using the SSI internal clock modifiers.
The variable baudclk_streams is introduced to keep track of the active
streams that are using the baudclock. This way we know if the baudclock
may be set and whether we may enable/disable the clock.
baudclock enable/disable is moved to hw_params()/hw_free(). This way we can
keep track of the baudclock in those two functions and avoid a running
clock while it is not used. As hw_params()/hw_free() may be called
multiple times for the same stream, we have to use baudclk_streams
variable to know whether we may enable/disable the clock.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Shengjiu Wang
Nicolin Chen
Mark Brown
Sean Cross
Nicolin Chen
Fabio Falzoi
Dan Carpenter
Fabio Estevam
Russell King - ARM Linux
Arnd Bergmann
Timur Tabi
Anssi Hannula
Guenter Roeck
Alexander Shiyan
Markus Pargmann