Commit 6fee37df authored by Mark Brown's avatar Mark Brown

Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and...

Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-mpc5200' into asoc-next
......@@ -24,6 +24,9 @@ The compatible list for this generic sound card currently:
"fsl,imx-audio-cs42888"
"fsl,imx-audio-cs427x"
(compatible with CS4271 and CS4272)
"fsl,imx-audio-wm8962"
(compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
......@@ -63,6 +66,12 @@ Optional properties:
- audio-asrc : The phandle of ASRC. It can be absent if there's no
need to add ASRC support via DPCM.
Optional unless SSI is selected as a CPU DAI:
- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
- mux-ext-port : The external port of the i.MX audio muxer
Example:
sound-cs42888 {
compatible = "fsl,imx-audio-cs42888";
......
......@@ -5,7 +5,7 @@ config SND_DAVINCI_SOC
config SND_EDMA_SOC
tristate "SoC Audio for Texas Instruments chips using eDMA"
depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI
depends on TI_EDMA
select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M here if you want audio support for TI SoC which uses eDMA.
......@@ -13,6 +13,7 @@ config SND_EDMA_SOC
- daVinci devices
- AM335x
- AM437x/AM438x
- DRA7xx family
config SND_DAVINCI_SOC_I2S
tristate
......
......@@ -77,6 +77,7 @@ struct davinci_mcasp {
u32 fifo_base;
struct device *dev;
struct snd_pcm_substream *substreams[2];
unsigned int dai_fmt;
/* McASP specific data */
int tdm_slots;
......@@ -398,6 +399,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
bool fs_pol_rising;
bool inv_fs = false;
if (!fmt)
return 0;
pm_runtime_get_sync(mcasp->dev);
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
......@@ -529,6 +533,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
}
mcasp->dai_fmt = fmt;
out:
pm_runtime_put(mcasp->dev);
return ret;
......@@ -1026,6 +1032,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
int period_size = params_period_size(params);
int ret;
ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
if (ret)
return ret;
/*
* If mcasp is BCLK master, and a BCLK divider was not provided by
* the machine driver, we need to calculate the ratio.
......@@ -1517,6 +1527,8 @@ static int mcasp_reparent_fck(struct platform_device *pdev)
if (!parent_name)
return 0;
dev_warn(&pdev->dev, "Update the bindings to use assigned-clocks!\n");
gfclk = clk_get(&pdev->dev, "fck");
if (IS_ERR(gfclk)) {
dev_err(&pdev->dev, "failed to get fck\n");
......
......@@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD
select SND_SOC_FSL_SSI
help
ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
and SGTL5000.
ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888,
CS4271, CS4272 and SGTL5000.
Say Y if you want to add support for Freescale Generic ASoC Sound Card.
endif # SND_IMX_SOC
......
......@@ -28,6 +28,8 @@
#include "../codecs/wm8962.h"
#include "../codecs/wm8960.h"
#define CS427x_SYSCLK_MCLK 0
#define RX 0
#define TX 1
......@@ -99,19 +101,26 @@ struct fsl_asoc_card_priv {
/**
* This dapm route map exsits for DPCM link only.
* The other routes shall go through Device Tree.
*
* Note: keep all ASRC routes in the second half
* to drop them easily for non-ASRC cases.
*/
static const struct snd_soc_dapm_route audio_map[] = {
{"CPU-Playback", NULL, "ASRC-Playback"},
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "CPU-Playback"},
{"ASRC-Capture", NULL, "CPU-Capture"},
{"CPU-Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"CPU-Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "CPU-Capture"},
};
static const struct snd_soc_dapm_route audio_map_ac97[] = {
{"AC97 Playback", NULL, "ASRC-Playback"},
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "AC97 Playback"},
{"ASRC-Capture", NULL, "AC97 Capture"},
{"AC97 Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"AC97 Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "AC97 Capture"},
};
/* Add all possible widgets into here without being redundant */
......@@ -528,6 +537,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.slot_width = 32;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
codec_dai_name = "cs4271-hifi";
priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
codec_dai_name = "sgtl5000";
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
......@@ -593,6 +606,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
/* Drop the second half of DAPM routes -- ASRC */
if (!asrc_pdev)
priv->card.num_dapm_routes /= 2;
memcpy(priv->dai_link, fsl_asoc_card_dai,
sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
......@@ -681,6 +698,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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-cs427x", },
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },
{ .compatible = "fsl,imx-audio-wm8960", },
......
......@@ -13,6 +13,7 @@
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <linux/delay.h>
#include <linux/time.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
......@@ -127,7 +128,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
mutex_unlock(&psc_dma->mutex);
msleep(1);
usleep_range(1000, 2000);
psc_ac97_warm_reset(ac97);
}
......
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