Merge series "Add audio driver base on rpmsg on i.MX platform" from Shengjiu Wang <shengjiu.wang@nxp.com>:
On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core,
Linux is running on A core, RTOS is running on M core.
The audio hardware device can be controlled by Cortex-M device,
So audio playback/capture can be handled by M core.
Rpmsg is the interface for sending and receiving msg to and from M
core, that we can create a virtual sound on Cortex-A core side.
A core will tell the Cortex-M core sound format/rate/channel,
where is the data buffer, what is the period size, when to start,
when to stop and when suspend or resume happen, each of this behavior
there is defined rpmsg command.
Especially we designed the low power audio case, that is to
allocate a large buffer and fill the data, then Cortex-A core can go
to sleep mode, Cortex-M core continue to play the sound, when the
buffer is consumed, Cortex-M core will trigger the Cortex-A core to
wakeup to fill data.
changes in v5:
- remove unneeded property in binding doc and driver
- update binding doc according to Rob's comments.
- Fix link issue reported by kernel test robot
changes in v4:
- remove the sound card node, merge the property to cpu dai node
according to Rob's comments.
- sound card device will be registered by cpu dai driver.
- Fix do_div issue reported by kernel test robot
changes in v3:
- add local refcount for clk enablement in hw_params()
- update the document according Rob's comments
changes in v2:
- update codes and comments according to Mark's comments
Shengjiu Wang (6):
ASoC: soc-component: Add snd_soc_pcm_component_ack
ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg
ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg audio device
ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel
ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg
ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg
.../devicetree/bindings/sound/fsl,rpmsg.yaml | 108 +++
include/sound/soc-component.h | 3 +
sound/soc/fsl/Kconfig | 30 +
sound/soc/fsl/Makefile | 6 +
sound/soc/fsl/fsl_rpmsg.c | 279 ++++++
sound/soc/fsl/fsl_rpmsg.h | 35 +
sound/soc/fsl/imx-audio-rpmsg.c | 140 +++
sound/soc/fsl/imx-pcm-rpmsg.c | 918 ++++++++++++++++++
sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++
sound/soc/fsl/imx-rpmsg.c | 150 +++
sound/soc/soc-component.c | 14 +
sound/soc/soc-pcm.c | 2 +
12 files changed, 2197 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
create mode 100644 sound/soc/fsl/fsl_rpmsg.c
create mode 100644 sound/soc/fsl/fsl_rpmsg.h
create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h
create mode 100644 sound/soc/fsl/imx-rpmsg.c
--
2.27.0
diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
index 90d9e9d..23d83fa 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt
@@ -42,6 +42,8 @@
"fsl,imx-audio-si476x"
+ "fsl,imx-audio-wm8958"
+
Required properties:
- compatible : Contains one of entries in the compatible list.
diff --git a/drivers/spi/spi-imx.c b/drivers/spi/spi-imx.c
index 5dc4ea4..9165807 100644
--- a/drivers/spi/spi-imx.c
+++ b/drivers/spi/spi-imx.c
@@ -1548,13 +1548,12 @@ static int spi_imx_slave_abort(struct spi_master *master)
static int spi_imx_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
- const struct of_device_id *of_id =
- of_match_device(spi_imx_dt_ids, &pdev->dev);
struct spi_master *master;
struct spi_imx_data *spi_imx;
struct resource *res;
int ret, irq, spi_drctl;
- const struct spi_imx_devtype_data *devtype_data = of_id->data;
+ const struct spi_imx_devtype_data *devtype_data =
+ of_device_get_match_data(&pdev->dev);
bool slave_mode;
u32 val;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 78609ab..e746da9 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -716,20 +716,38 @@ struct snd_soc_dai_link {
struct snd_soc_dobj dobj; /* For topology */
#endif
};
+
+static inline struct snd_soc_dai_link_component*
+asoc_link_to_cpu(struct snd_soc_dai_link *link, int n) {
+ return &(link)->cpus[n];
+}
+
+static inline struct snd_soc_dai_link_component*
+asoc_link_to_codec(struct snd_soc_dai_link *link, int n) {
+ return &(link)->codecs[n];
+}
+
+static inline struct snd_soc_dai_link_component*
+asoc_link_to_platform(struct snd_soc_dai_link *link, int n) {
+ return &(link)->platforms[n];
+}
+
#define for_each_link_codecs(link, i, codec) \
for ((i) = 0; \
- ((i) < link->num_codecs) && ((codec) = &link->codecs[i]); \
+ ((i) < link->num_codecs) && \
+ ((codec) = asoc_link_to_codec(link, i)); \
(i)++)
#define for_each_link_platforms(link, i, platform) \
for ((i) = 0; \
((i) < link->num_platforms) && \
- ((platform) = &link->platforms[i]); \
+ ((platform) = asoc_link_to_platform(link, i)); \
(i)++)
#define for_each_link_cpus(link, i, cpu) \
for ((i) = 0; \
- ((i) < link->num_cpus) && ((cpu) = &link->cpus[i]); \
+ ((i) < link->num_cpus) && \
+ ((cpu) = asoc_link_to_cpu(link, i)); \
(i)++)
/*
@@ -1262,13 +1280,17 @@ int snd_soc_fixup_dai_links_platform_name(struct snd_soc_card *card,
/* set platform name for each dailink */
for_each_card_prelinks(card, i, dai_link) {
- name = devm_kstrdup(card->dev, platform_name, GFP_KERNEL);
- if (!name)
- return -ENOMEM;
+ /* only single platform is supported for now */
+ if (dai_link->num_platforms != 1)
+ return -EINVAL;
if (!dai_link->platforms)
return -EINVAL;
+ name = devm_kstrdup(card->dev, platform_name, GFP_KERNEL);
+ if (!name)
+ return -ENOMEM;
+
/* only single platform is supported for now */
dai_link->platforms->name = name;
}
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 746c829..b1e43c2 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -202,6 +202,15 @@ SOC_SINGLE("Ramp Up Switch", MAX98373_R203F_AMP_DSP_CFG,
MAX98373_AMP_DSP_CFG_RMP_UP_SHIFT, 1, 0),
SOC_SINGLE("Ramp Down Switch", MAX98373_R203F_AMP_DSP_CFG,
MAX98373_AMP_DSP_CFG_RMP_DN_SHIFT, 1, 0),
+/* Speaker Amplifier Overcurrent Automatic Restart Enable */
+SOC_SINGLE("OVC Autorestart Switch", MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG,
+ MAX98373_OVC_AUTORESTART_SHIFT, 1, 0),
+/* Thermal Shutdown Automatic Restart Enable */
+SOC_SINGLE("THERM Autorestart Switch", MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG,
+ MAX98373_THERM_AUTORESTART_SHIFT, 1, 0),
+/* Clock Monitor Automatic Restart Enable */
+SOC_SINGLE("CMON Autorestart Switch", MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG,
+ MAX98373_CMON_AUTORESTART_SHIFT, 1, 0),
SOC_SINGLE("CLK Monitor Switch", MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG,
MAX98373_CLOCK_MON_SHIFT, 1, 0),
SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
@@ -390,6 +399,11 @@ static int max98373_probe(struct snd_soc_component *component)
MAX98373_R2021_PCM_TX_HIZ_EN_2,
1 << (max98373->i_slot - 8), 0);
+ /* enable auto restart function by default */
+ regmap_write(max98373->regmap,
+ MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG,
+ 0xF);
+
/* speaker feedback slot configuration */
regmap_write(max98373->regmap,
MAX98373_R2023_PCM_TX_SRC_2,
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
index 71f5a52..73a2cf6 100644
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -195,6 +195,9 @@
#define MAX98373_LIMITER_EN_SHIFT (0)
/* MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG */
+#define MAX98373_OVC_AUTORESTART_SHIFT (3)
+#define MAX98373_THERM_AUTORESTART_SHIFT (2)
+#define MAX98373_CMON_AUTORESTART_SHIFT (1)
#define MAX98373_CLOCK_MON_SHIFT (0)
/* MAX98373_R20FF_GLOBAL_SHDN */
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 10a7b77..9238f12 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -669,8 +669,23 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w,
struct snd_soc_component *component =
snd_soc_dapm_to_component(w->dapm);
struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component);
+ unsigned int ret, ret2;
switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ ret = snd_soc_component_read(component, RT1015_CLK_DET);
+ ret2 = snd_soc_component_read(component, RT1015_SPK_DC_DETECT1);
+ if (!((ret >> 15) & 0x1)) {
+ snd_soc_component_update_bits(component, RT1015_CLK_DET,
+ RT1015_EN_BCLK_DET_MASK, RT1015_EN_BCLK_DET);
+ dev_dbg(component->dev, "BCLK Detection Enabled.\n");
+ }
+ if (!((ret2 >> 12) & 0x1)) {
+ snd_soc_component_update_bits(component, RT1015_SPK_DC_DETECT1,
+ RT1015_EN_CLA_D_DC_DET_MASK, RT1015_EN_CLA_D_DC_DET);
+ dev_dbg(component->dev, "Class-D DC Detection Enabled.\n");
+ }
+ break;
case SND_SOC_DAPM_POST_PMU:
if (rt1015->hw_config == RT1015_HW_28)
schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10));
@@ -690,7 +705,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = {
r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0,
- rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU),
+ rt1015_amp_drv_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_OUTPUT("SPO"),
};
diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h
index 2aeaf65..1434453 100644
--- a/sound/soc/codecs/rt1015.h
+++ b/sound/soc/codecs/rt1015.h
@@ -209,6 +209,11 @@
#define RT1015_PLL_K_MASK (RT1015_PLL_K_MAX)
#define RT1015_PLL_K_SFT 0
+/* 0x0020 */
+#define RT1015_EN_BCLK_DET_MASK (0x1 << 15)
+#define RT1015_EN_BCLK_DET (0x1 << 15)
+#define RT1015_DIS_BCLK_DET (0x0 << 15)
+
/* 0x007a */
#define RT1015_ID_MASK 0xff
#define RT1015_ID_VERA 0x0
@@ -374,6 +379,11 @@
#define RT1015_PWR_SWR (0x1 << 12)
#define RT1015_PWR_SWR_BIT 12
+/* 0x0519 */
+#define RT1015_EN_CLA_D_DC_DET_MASK (0x1 << 12)
+#define RT1015_EN_CLA_D_DC_DET (0x1 << 12)
+#define RT1015_DIS_CLA_D_DC_DET (0x0 << 12)
+
/* 0x1300 */
#define RT1015_PWR_CLSD (0x1 << 12)
#define RT1015_PWR_CLSD_BIT 12
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index df35151..847ca16 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -608,10 +608,6 @@ static const int bclk_divs[] = {
* - lrclk = sysclk / dac_divs
* - 10 * bclk = sysclk / bclk_divs
*
- * If we cannot find an exact match for (sysclk, lrclk, bclk)
- * triplet, we relax the bclk such that bclk is chosen as the
- * closest available frequency greater than expected bclk.
- *
* @wm8960: codec private data
* @mclk: MCLK used to derive sysclk
* @sysclk_idx: sysclk_divs index for found sysclk
@@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk,
{
int sysclk, bclk, lrclk;
int i, j, k;
- int diff, closest = mclk;
+ int diff;
/* marker for no match */
*bclk_idx = -1;
@@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk,
*bclk_idx = k;
break;
}
- if (diff > 0 && closest > diff) {
- *sysclk_idx = i;
- *dac_idx = j;
- *bclk_idx = k;
- closest = diff;
- }
}
if (k != ARRAY_SIZE(bclk_divs))
break;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index c71c602..0917d65 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -310,6 +310,8 @@
select SND_SOC_FSL_ESAI
select SND_SOC_FSL_SAI
select SND_SOC_FSL_SSI
+ select SND_SOC_WM8994
+ select MFD_WM8994
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,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index f62f81c..c62bfd1 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -25,6 +25,7 @@
#include "../codecs/sgtl5000.h"
#include "../codecs/wm8962.h"
#include "../codecs/wm8960.h"
+#include "../codecs/wm8994.h"
#define CS427x_SYSCLK_MCLK 0
@@ -37,12 +38,14 @@
/**
* struct codec_priv - CODEC private data
* @mclk_freq: Clock rate of MCLK
+ * @free_freq: Clock rate of MCLK for hw_free()
* @mclk_id: MCLK (or main clock) id for set_sysclk()
* @fll_id: FLL (or secordary clock) id for set_sysclk()
* @pll_id: PLL id for set_pll()
*/
struct codec_priv {
unsigned long mclk_freq;
+ unsigned long free_freq;
u32 mclk_id;
u32 fll_id;
u32 pll_id;
@@ -235,10 +238,10 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
priv->streams &= ~BIT(substream->stream);
if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
- /* Force freq to be 0 to avoid error message in codec */
+ /* Force freq to be free_freq to avoid error message in codec */
ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
codec_priv->mclk_id,
- 0,
+ codec_priv->free_freq,
SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "failed to switch away from FLL: %d\n", ret);
@@ -665,6 +668,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
priv->card.dapm_routes = audio_map_rx;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
+ codec_dai_name = "wm8994-aif1";
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1;
+ priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1;
+ priv->codec_priv.pll_id = WM8994_FLL1;
+ priv->codec_priv.free_freq = priv->codec_priv.mclk_freq;
+ priv->card.dapm_routes = NULL;
+ priv->card.num_dapm_routes = 0;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
@@ -882,6 +894,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-mqs", },
{ .compatible = "fsl,imx-audio-wm8524", },
{ .compatible = "fsl,imx-audio-si476x", },
+ { .compatible = "fsl,imx-audio-wm8958", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);