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ASoC: fsl-asoc-card: add AC'97 support 

Add AC'97 support to fsl-asoc-card using generic
ASoC AC'97 CODEC.

The SSI controller will silently enable any TX
AC'97 slots that have their bits set in SLOTREQ
received from CODEC and then will redirect some
of playback samples there.

That's why it is important to make sure that
any of CODEC playback slots that can pull samples
are set to slots 3/4 (standard PCM playback slots).
Currently, this applies to S/PDIF slots as they
were seen to pull samples sometimes even with
S/PDIF output being disabled.

Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
hifive-unleashed-5.1
Maciej S. Szmigiero 2015-09-19 02:00:25 +02:00 committed by Mark Brown
parent 6ff33f3902
commit 50760cad9d
2 changed files with 116 additions and 34 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
View File

@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit
from the simplification of a new card support and the capability of the wide
sample rates support through ASRC.
Note: The card is initially designed for those sound cards who use I2S and
PCM DAI formats. However, it'll be also possible to support those non
I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long
as the driver has been properly upgraded.
Note: The card is initially designed for those sound cards who use AC'97, I2S
and PCM DAI formats. However, it'll be also possible to support those non
AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
long as the driver has been properly upgraded.
The compatible list for this generic sound card currently:
"fsl,imx-audio-ac97"
"fsl,imx-audio-cs42888"
"fsl,imx-audio-wm8962"

140
sound/soc/fsl/fsl-asoc-card.c
View File

@ -14,6 +14,9 @@
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/of_platform.h>
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
#include <sound/ac97_codec.h>
#endif
#include <sound/pcm_params.h>
#include <sound/soc.h>
@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
SND_SOC_DAPM_MIC("DMIC", NULL),
};
static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
{
return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
}
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
* set_bias_level(), bypass the remaining settings in hw_params().
* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
*/
if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
if ((priv->card.set_bias_level &&
priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
ext_port--;
/*
* Use asynchronous mode (6 wires) for all cases.
* Use asynchronous mode (6 wires) for all cases except AC97.
* If only 4 wires are needed, just set SSI into
* synchronous mode and enable 4 PADs in IOMUX.
*/
@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
default:
return -EINVAL;
if (!fsl_asoc_card_is_ac97(priv))
return -EINVAL;
}
if (fsl_asoc_card_is_ac97(priv)) {
int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_TFSDIR;
}
/* Asynchronous mode can not be set along with RCLKDIR */
ret = imx_audmux_v2_configure_port(int_port, 0,
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
ret = imx_audmux_v2_configure_port(int_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
return ret;
}
ret = imx_audmux_v2_configure_port(ext_port, 0,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
ret = imx_audmux_v2_configure_port(ext_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
struct device *dev = card->dev;
int ret;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
struct snd_soc_codec *codec = card->rtd[0].codec;
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
/*
* Use slots 3/4 for S/PDIF so SSI won't try to enable
* other slots and send some samples there
* due to SLOTREQ bits for S/PDIF received from codec
*/
snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
#endif
return 0;
}
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
if (ret) {
@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
struct i2c_client *codec_dev;
struct clk *codec_clk;
const char *codec_dai_name;
u32 width;
int ret;
@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Give a chance to old DT binding */
if (!cpu_np)
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (!cpu_np || !codec_np) {
dev_err(&pdev->dev, "phandle missing or invalid\n");
if (!cpu_np) {
dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
goto fail;
}
codec_dev = of_find_i2c_device_by_node(codec_np);
if (!codec_dev) {
dev_err(&pdev->dev, "failed to find codec platform device\n");
ret = -EINVAL;
goto fail;
}
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (codec_np)
codec_dev = of_find_i2c_device_by_node(codec_np);
else
codec_dev = NULL;
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
asrc_pdev = of_find_device_by_node(asrc_np);
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
codec_clk = clk_get(&codec_dev->dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
clk_put(codec_clk);
if (codec_dev) {
struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
clk_put(codec_clk);
}
}
/* Default sample rate and format, will be updated in hw_params() */
@ -486,11 +533,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
return -EINVAL;
}
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
dev_err(&pdev->dev, "failed to find codec device\n");
ret = -EINVAL;
goto asrc_fail;
}
/* Common settings for corresponding Freescale CPU DAI driver */
if (strstr(cpu_np->name, "ssi")) {
/* Only SSI needs to configure AUDMUX */
@ -507,7 +564,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
sprintf(priv->name, "%s-audio", codec_dev->name);
snprintf(priv->name, sizeof(priv->name), "%s-audio",
fsl_asoc_card_is_ac97(priv) ? "ac97" :
codec_dev->name);
/* Initialize sound card */
priv->pdev = pdev;
@ -531,8 +590,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Normal DAI Link */
priv->dai_link[0].cpu_of_node = cpu_np;
priv->dai_link[0].codec_of_node = codec_np;
priv->dai_link[0].codec_dai_name = codec_dai_name;
if (!fsl_asoc_card_is_ac97(priv))
priv->dai_link[0].codec_of_node = codec_np;
else {
u32 idx;
ret = of_property_read_u32(cpu_np, "cell-index", &idx);
if (ret) {
dev_err(&pdev->dev,
"cannot get CPU index property\n");
goto asrc_fail;
}
priv->dai_link[0].codec_name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
"ac97-codec.%u",
(unsigned int)idx);
}
priv->dai_link[0].platform_of_node = cpu_np;
priv->dai_link[0].dai_fmt = priv->dai_fmt;
priv->card.num_links = 1;
@ -543,6 +620,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_link[1].platform_of_node = asrc_np;
priv->dai_link[2].codec_dai_name = codec_dai_name;
priv->dai_link[2].codec_of_node = codec_np;
priv->dai_link[2].codec_name =
priv->dai_link[0].codec_name;
priv->dai_link[2].cpu_of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->card.num_links = 3;
@ -578,14 +657,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
asrc_fail:
of_node_put(asrc_np);
fail:
of_node_put(codec_np);
fail:
of_node_put(cpu_np);
return ret;
}
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-ac97", },
{ .compatible = "fsl,imx-audio-cs42888", },
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },