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Commit Graph

554 Commits

Author SHA1 Message Date
Mark Brown
bdf20b4291 Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus 2014-10-08 16:44:50 +01:00
Takashi Iwai
960baba41f ASoC: imx-es8328: Fix of_node_put() call with uninitialized object
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>
2014-10-07 23:51:39 +01:00
Takashi Iwai
5c4c99f322 ASoC: imx-es8328: Fix missing return code in imx_es8328_probe()
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>
2014-10-07 18:10:32 +01:00
Mark Brown
248519c00e Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/spdif', 'asoc/topic/ssm2602' and 'asoc/topic/ssm4567' into asoc-next 2014-10-06 12:49:05 +01:00
Mark Brown
57b027f697 Merge remote-tracking branches 'asoc/topic/fsl-easi', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-ssi' and 'asoc/topic/intel' into asoc-next 2014-10-06 12:48:59 +01:00
Mark Brown
565fefdf31 Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/dmic', 'asoc/topic/drivers', 'asoc/topic/es8328' and 'asoc/topic/fsl' into asoc-next 2014-10-06 12:48:57 +01:00
Mark Brown
7ddb870b78 Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next 2014-10-06 12:48:53 +01:00
Tomeu Vizoso
58a9014ae6 ASoC: fsl_spdif: Remove unused includes of linux/clk-private.h
Signed-off-by: Tomeu Vizoso <tomeu.vizoso@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2014-10-03 17:19:32 +01:00
Fabio Estevam
c05a11f7b8 ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD
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>
2014-10-01 17:07:05 +01:00
Fabio Estevam
ece1e49996 ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property
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>
2014-09-30 13:43:03 +01:00
Mark Brown
82b925c405 Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/core', 'asoc/fix/fsl-ssi' and 'asoc/fix/rt286' into asoc-linus 2014-09-28 12:25:12 +01:00
Michael Trimarchi
85151461f1 ASoC: fsl_ssi: fix kernel panic in probe function
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>
2014-09-22 18:58:48 -07:00
Shengjiu Wang
f4a43caba7 ASoC: fsl_ssi: refine ipg clock usage in this module
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>
2014-09-17 11:30:15 -07:00
Xiubo Li
eadb0019d2 ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'.
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>
2014-09-01 16:36:42 +01:00
Mark Brown
025b78b809 Merge branch 'topic/fsl' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-sai 2014-09-01 16:36:34 +01:00
Mark Brown
5b87d31309 Merge branch 'topic/fsl' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-esai 2014-09-01 10:49:32 +01:00
Xiubo Li
014fd22ef9 ASoC: fsl-sai: Convert to use regmap framework's endianness method.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-27 19:19:29 +01:00
Xiubo Li
664915074e ASoC: fsl-spdif: Convert to use regmap framework's endianness method.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-27 19:19:29 +01:00
Xiubo Li
92bd0334b2 ASoC: fsl-esai: Convert to use regmap framework's endianness method.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-27 19:19:28 +01:00
Xiubo Li
bf16d88326 ASoC: fsl-asrc: Convert to use regmap framework's endianness method.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-27 19:19:20 +01:00
Shengjiu Wang
38c6e4bb67 ASoC: fsl-asoc-card: move 'config SND_SOC_FSL_ASOC_CARD' to 'if SND_IMX_SOC'
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>
2014-08-19 11:11:50 -05:00
Sean Cross
cdec729765 ASoC: fsl: Fix building of imx-es8328 on PPC
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>
2014-08-19 10:43:45 -05:00
Shengjiu Wang
499898d66d ASoC: fsl: fsl-asoc-card: Select SND_SOC_IMX_AUDMUX
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>
2014-08-18 09:54:04 -05:00
Shengjiu Wang
5f37671e00 ASoC: fsl-asoc-card: Fix build warning for maybe-uninitialized
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>
2014-08-18 09:52:51 -05:00
Nicolin Chen
855675f6e6 ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode
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>
2014-08-16 17:30:45 -05:00
Mark Brown
6be1f475e0 Merge branch 'fix/fsl-esai' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-esai 2014-08-16 17:22:36 -05:00
Sean Cross
7e7292dba2 ASoC: fsl: add imx-es8328 machine driver
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>
2014-08-16 17:18:07 -05:00
Nicolin Chen
ce7344a4eb ASoC: fsl_sai: Make Synchronous and Asynchronous modes exclusive
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>
2014-08-16 17:06:23 -05:00
Nicolin Chen
08fdf65e37 ASoC: fsl_sai: Add asynchronous mode support
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>
2014-08-16 17:06:23 -05:00
Nicolin Chen
af96ff5b74 ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode
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>
2014-08-16 17:06:23 -05:00
Nicolin Chen
376d1a92ca ASoC: fsl_sai: Initialize with software reset
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>
2014-08-16 17:06:23 -05:00
Shengjiu Wang
de0d712a6d ASoC: fsl_esai: refine esai for TDM support
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>
2014-08-16 17:06:15 -05:00
Nicolin Chen
708b4351f0 ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support
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>
2014-08-16 17:03:50 -05:00
Shengjiu Wang
d177143c36 ASoC: fsl_esai: refine esai for TDM support
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>
2014-08-13 19:50:21 +01:00
Shengjiu Wang
769091ee18 ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() support
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>
2014-08-13 19:50:00 +01:00
Fabio Falzoi
cf4f7fc3e7 ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS
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>
2014-08-07 18:45:38 +01:00
Mark Brown
e7177999dc Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-spdif' and 'asoc/topic/imx-audmux' into asoc-next 2014-08-04 16:31:40 +01:00
Mark Brown
3674b710a7 Merge remote-tracking branch 'asoc/fix/fsl-sai' into asoc-linus 2014-08-04 16:31:12 +01:00
Mark Brown
e5f89768e9 ASoC: imx-audmux: Use uintptr_t for port numbers
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>
2014-08-04 16:30:24 +01:00
Nicolin Chen
4e13eb7221 ASoC: fsl_asrc: Don't access members of config before checking it
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 3117bb3109
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>
2014-08-04 15:48:41 +01:00
Nicolin Chen
6ccf62c7be ASoC: fsl_sarc_dma: Check pair before using it
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>
2014-08-04 15:48:41 +01:00
Nicolin Chen
e365500459 ASoC: fsl_ssi: Add stream names for DPCM usage
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>
2014-07-31 20:55:08 +01:00
Nicolin Chen
756409320b ASoC: fsl_spdif: Add stream names for DPCM usage
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>
2014-07-31 20:55:04 +01:00
Nicolin Chen
20d5b76fb2 ASoC: fsl_sai: Add stream names for DPCM usage
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>
2014-07-31 20:55:04 +01:00
Nicolin Chen
74ccb27c35 ASoC: fsl_esai: Add stream names for DPCM usage
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>
2014-07-31 20:55:04 +01:00
Dan Carpenter
d387dd08e4 ASoC: fsl_asrc: fix an error code in fsl_asrc_probe()
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>
2014-07-31 20:23:24 +01:00
Nicolin Chen
d526416c4f ASoC: fsl_asrc: Fix sparse warnings in FSL_ASRC_FORMATS due to typo
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>
2014-07-31 20:22:33 +01:00
Fabio Estevam
bdb9eb4967 ASoC: fsl: fsl_asrc: Select SND_SOC_GENERIC_DMAENGINE_PCM
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>
2014-07-31 00:04:23 +01:00
Fabio Estevam
d3dacda939 ASoC: fsl_asrc: Use 'ifdef' for config options
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>
2014-07-29 20:15:47 +01:00
Nicolin Chen
3117bb3109 ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers
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>
2014-07-29 19:22:49 +01:00