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authorTakashi Iwai <tiwai@suse.de>2022-10-03 17:30:42 +0300
committerTakashi Iwai <tiwai@suse.de>2022-10-03 17:30:42 +0300
commit86a4d29e75540e20f991e72f17aa51d0e775a397 (patch)
tree87d99ee3855d97da60074fdc9c0cdd63dd2da66b /sound/soc/fsl
parent02f2e785c4834828876a4701926416157dfd7b26 (diff)
parentf0c8d7468af0001b80b0c86802ee28063f800987 (diff)
downloadlinux-86a4d29e75540e20f991e72f17aa51d0e775a397.tar.xz
Merge tag 'asoc-v6.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.1 This has been a very quiet release for the core but quite a busy one for drivers with a big crop of new drivers and lots of feature additions and fixes to existing ones: - A new string helper parse_int_array_user(). - Improvements to the SOF IPC4 code, especially around trace. - Support for AMD Rembrant DSPs, AMD Pink Sardine ACP 6.2, Apple Silcon systems, Everest ES8326, Intel Sky Lake and Kaby Lake, MediaTek MT8186 support, NXP i.MX8ULP DSPs, Qualcomm SC8280XP, SM8250 and SM8450 and Texas Instruments SRC4392 There is a conflict with the conversion of I2C remove functions to void in the cs42l42 driver which is fairly straightforward to resolve but should be highlighted to Linus.
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c8
-rw-r--r--sound/soc/fsl/fsl_asrc.c29
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c3
-rw-r--r--sound/soc/fsl/fsl_sai.c55
-rw-r--r--sound/soc/fsl/fsl_sai.h3
-rw-r--r--sound/soc/fsl/fsl_spdif.c21
-rw-r--r--sound/soc/fsl/fsl_utils.c2
-rw-r--r--sound/soc/fsl/imx-pcm.h9
-rw-r--r--sound/soc/fsl/imx-rpmsg.c29
9 files changed, 130 insertions, 29 deletions
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 8b61582753c8..9af4c4a35eb1 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -86,7 +86,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
int ret;
int int_port = 0, ext_port;
struct device_node *np = pdev->dev.of_node;
- struct device_node *ssi_np = NULL, *codec_np = NULL;
+ struct device_node *ssi_np = NULL, *codec_np = NULL, *tmp_np = NULL;
eukrea_tlv320.dev = &pdev->dev;
if (np) {
@@ -143,7 +143,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
}
if (machine_is_eukrea_cpuimx27() ||
- of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) {
+ (tmp_np = of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux"))) {
imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
IMX_AUDMUX_V1_PCR_SYN |
IMX_AUDMUX_V1_PCR_TFSDIR |
@@ -158,10 +158,11 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
IMX_AUDMUX_V1_PCR_SYN |
IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
);
+ of_node_put(tmp_np);
} else if (machine_is_eukrea_cpuimx25sd() ||
machine_is_eukrea_cpuimx35sd() ||
machine_is_eukrea_cpuimx51sd() ||
- of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) {
+ (tmp_np = of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux"))) {
if (!np)
ext_port = machine_is_eukrea_cpuimx25sd() ?
4 : 3;
@@ -178,6 +179,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev)
IMX_AUDMUX_V2_PTCR_SYN,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)
);
+ of_node_put(tmp_np);
} else {
if (np) {
/* The eukrea,asoc-tlv320 driver was explicitly
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index aa5edf32d988..936aef5d2767 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -20,6 +20,7 @@
#define IDEAL_RATIO_DECIMAL_DEPTH 26
#define DIVIDER_NUM 64
+#define INIT_RETRY_NUM 50
#define pair_err(fmt, ...) \
dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
@@ -27,6 +28,9 @@
#define pair_dbg(fmt, ...) \
dev_dbg(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+#define pair_warn(fmt, ...) \
+ dev_warn(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+
/* Corresponding to process_option */
static unsigned int supported_asrc_rate[] = {
5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
@@ -579,7 +583,7 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair)
{
struct fsl_asrc *asrc = pair->asrc;
enum asrc_pair_index index = pair->index;
- int reg, retry = 10, i;
+ int reg, retry = INIT_RETRY_NUM, i;
/* Enable the current pair */
regmap_update_bits(asrc->regmap, REG_ASRCTR,
@@ -592,6 +596,10 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair)
reg &= ASRCFG_INIRQi_MASK(index);
} while (!reg && --retry);
+ /* NOTE: Doesn't treat initialization timeout as an error */
+ if (!retry)
+ pair_warn("initialization isn't finished\n");
+
/* Make the input fifo to ASRC STALL level */
regmap_read(asrc->regmap, REG_ASRCNCR, &reg);
for (i = 0; i < pair->channels * 4; i++)
@@ -1257,6 +1265,7 @@ static int fsl_asrc_runtime_resume(struct device *dev)
{
struct fsl_asrc *asrc = dev_get_drvdata(dev);
struct fsl_asrc_priv *asrc_priv = asrc->private;
+ int reg, retry = INIT_RETRY_NUM;
int i, ret;
u32 asrctr;
@@ -1295,6 +1304,24 @@ static int fsl_asrc_runtime_resume(struct device *dev)
regmap_update_bits(asrc->regmap, REG_ASRCTR,
ASRCTR_ASRCEi_ALL_MASK, asrctr);
+ /* Wait for status of initialization for all enabled pairs */
+ do {
+ udelay(5);
+ regmap_read(asrc->regmap, REG_ASRCFG, &reg);
+ reg = (reg >> ASRCFG_INIRQi_SHIFT(0)) & 0x7;
+ } while ((reg != ((asrctr >> ASRCTR_ASRCEi_SHIFT(0)) & 0x7)) && --retry);
+
+ /*
+ * NOTE: Doesn't treat initialization timeout as an error
+ * Some of the pairs may success, then still can continue.
+ */
+ if (!retry) {
+ for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) {
+ if ((asrctr & ASRCTR_ASRCEi_MASK(i)) && !(reg & (1 << i)))
+ dev_warn(dev, "Pair %c initialization isn't finished\n", 'A' + i);
+ }
+ }
+
return 0;
disable_asrck_clk:
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 12ddf2320f2d..3b81a465814a 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -139,7 +139,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
struct dma_chan *tmp_chan = NULL, *be_chan = NULL;
struct snd_soc_component *component_be = NULL;
struct fsl_asrc *asrc = pair->asrc;
- struct dma_slave_config config_fe, config_be;
+ struct dma_slave_config config_fe = {}, config_be = {};
struct sdma_peripheral_config audio_config;
enum asrc_pair_index index = pair->index;
struct device *dev = component->dev;
@@ -183,7 +183,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
return -EINVAL;
}
- memset(&config_fe, 0, sizeof(config_fe));
ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe);
if (ret) {
dev_err(dev, "failed to prepare DMA config for Front-End\n");
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index d430eece1d6b..81f89f6767a2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -114,11 +114,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid)
if (flags & FSL_SAI_CSR_SEF)
dev_dbg(dev, "isr: Tx Frame sync error detected\n");
- if (flags & FSL_SAI_CSR_FEF) {
+ if (flags & FSL_SAI_CSR_FEF)
dev_dbg(dev, "isr: Transmit underrun detected\n");
- /* FIFO reset for safety */
- xcsr |= FSL_SAI_CSR_FR;
- }
if (flags & FSL_SAI_CSR_FWF)
dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n");
@@ -148,11 +145,8 @@ irq_rx:
if (flags & FSL_SAI_CSR_SEF)
dev_dbg(dev, "isr: Rx Frame sync error detected\n");
- if (flags & FSL_SAI_CSR_FEF) {
+ if (flags & FSL_SAI_CSR_FEF)
dev_dbg(dev, "isr: Receive overflow detected\n");
- /* FIFO reset for safety */
- xcsr |= FSL_SAI_CSR_FR;
- }
if (flags & FSL_SAI_CSR_FWF)
dev_dbg(dev, "isr: Enabled receive FIFO is full\n");
@@ -533,14 +527,17 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
u32 slot_width = word_width;
int adir = tx ? RX : TX;
u32 pins, bclk;
+ u32 watermark;
int ret, i;
- if (sai->slots)
- slots = sai->slots;
-
if (sai->slot_width)
slot_width = sai->slot_width;
+ if (sai->slots)
+ slots = sai->slots;
+ else if (sai->bclk_ratio)
+ slots = sai->bclk_ratio / slot_width;
+
pins = DIV_ROUND_UP(channels, slots);
/*
@@ -625,7 +622,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
FSL_SAI_CR5_FBT_MASK, val_cr5);
}
- if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1)
+ /*
+ * Combine mode has limation:
+ * - Can't used for singel dataline/FIFO case except the FIFO0
+ * - Can't used for multi dataline/FIFO case except the enabled FIFOs
+ * are successive and start from FIFO0
+ *
+ * So for common usage, all multi fifo case disable the combine mode.
+ */
+ if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma)
regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
FSL_SAI_CR4_FCOMB_MASK, 0);
else
@@ -636,6 +641,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) +
dl_cfg[dl_cfg_idx].start_off[tx] * 0x4;
+ if (sai->is_multi_fifo_dma) {
+ sai->audio_config[tx].words_per_fifo = min(slots, channels);
+ if (tx) {
+ sai->audio_config[tx].n_fifos_dst = pins;
+ sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx];
+ } else {
+ sai->audio_config[tx].n_fifos_src = pins;
+ sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx];
+ }
+ dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins;
+ dma_params->peripheral_config = &sai->audio_config[tx];
+ dma_params->peripheral_size = sizeof(sai->audio_config[tx]);
+
+ watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) :
+ (dma_params->maxburst - 1);
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs),
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+ watermark);
+ }
+
/* Find a proper tcre setting */
for (i = 0; i < sai->soc_data->pins; i++) {
trce_mask = (1 << (i + 1)) - 1;
@@ -1263,6 +1288,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
char tmp[8];
int irq, ret, i;
int index;
+ u32 dmas[4];
sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
@@ -1319,6 +1345,11 @@ static int fsl_sai_probe(struct platform_device *pdev)
fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk,
&sai->pll11k_clk);
+ /* Use Multi FIFO mode depending on the support from SDMA script */
+ ret = of_property_read_u32_array(np, "dmas", dmas, 4);
+ if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI)
+ sai->is_multi_fifo_dma = true;
+
/* read dataline mask for rx and tx*/
ret = fsl_sai_read_dlcfg(sai);
if (ret < 0) {
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 17956b5731dc..697f6690068c 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -6,6 +6,7 @@
#ifndef __FSL_SAI_H
#define __FSL_SAI_H
+#include <linux/dma/imx-dma.h>
#include <sound/dmaengine_pcm.h>
#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
@@ -281,6 +282,7 @@ struct fsl_sai {
bool is_lsb_first;
bool is_dsp_mode;
bool is_pdm_mode;
+ bool is_multi_fifo_dma;
bool synchronous[2];
struct fsl_sai_dl_cfg *dl_cfg;
unsigned int dl_cfg_cnt;
@@ -300,6 +302,7 @@ struct fsl_sai {
struct pm_qos_request pm_qos_req;
struct pinctrl *pinctrl;
struct pinctrl_state *pins_state;
+ struct sdma_peripheral_config audio_config[2];
};
#define TX 1
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 7fc1c96929bb..275aba8e0c46 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -44,6 +44,8 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
#define DEFAULT_RXCLK_SRC 1
+#define RX_SAMPLE_RATE_KCONTROL "RX Sample Rate"
+
/**
* struct fsl_spdif_soc_data: soc specific data
*
@@ -98,6 +100,8 @@ struct spdif_mixer_control {
* @soc: SPDIF soc data
* @fsl_spdif_control: SPDIF control data
* @cpu_dai_drv: cpu dai driver
+ * @snd_card: sound card pointer
+ * @rxrate_kcontrol: kcontrol for RX Sample Rate
* @pdev: platform device pointer
* @regmap: regmap handler
* @dpll_locked: dpll lock flag
@@ -122,6 +126,8 @@ struct fsl_spdif_priv {
const struct fsl_spdif_soc_data *soc;
struct spdif_mixer_control fsl_spdif_control;
struct snd_soc_dai_driver cpu_dai_drv;
+ struct snd_card *snd_card;
+ struct snd_kcontrol *rxrate_kcontrol;
struct platform_device *pdev;
struct regmap *regmap;
bool dpll_locked;
@@ -226,6 +232,12 @@ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
locked ? "locked" : "loss lock");
spdif_priv->dpll_locked = locked ? true : false;
+
+ if (spdif_priv->snd_card && spdif_priv->rxrate_kcontrol) {
+ snd_ctl_notify(spdif_priv->snd_card,
+ SNDRV_CTL_EVENT_MASK_VALUE,
+ &spdif_priv->rxrate_kcontrol->id);
+ }
}
/* Receiver found illegal symbol interrupt handler */
@@ -1197,7 +1209,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
/* DPLL lock info get controller */
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
- .name = "RX Sample Rate",
+ .name = RX_SAMPLE_RATE_KCONTROL,
.access = SNDRV_CTL_ELEM_ACCESS_READ |
SNDRV_CTL_ELEM_ACCESS_VOLATILE,
.info = fsl_spdif_rxrate_info,
@@ -1251,6 +1263,13 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
snd_soc_add_dai_controls(dai, fsl_spdif_ctrls_rcm,
ARRAY_SIZE(fsl_spdif_ctrls_rcm));
+ spdif_private->snd_card = dai->component->card->snd_card;
+ spdif_private->rxrate_kcontrol = snd_soc_card_get_kcontrol(dai->component->card,
+ RX_SAMPLE_RATE_KCONTROL);
+ if (!spdif_private->rxrate_kcontrol)
+ dev_err(&spdif_private->pdev->dev, "failed to get %s kcontrol\n",
+ RX_SAMPLE_RATE_KCONTROL);
+
/*Clear the val bit for Tx*/
regmap_update_bits(spdif_private->regmap, REG_SPDIF_SCR,
SCR_VAL_MASK, SCR_VAL_CLEAR);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index d0fc430f7033..a5ab27c2f711 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -124,7 +124,7 @@ void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk,
{
struct clk *p, *pll = NULL, *npll = NULL;
bool reparent = false;
- int ret = 0;
+ int ret;
if (!clk || !pll8k_clk || !pll11k_clk)
return;
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 06b25f4b26b6..ac5f57c3cc55 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -18,15 +18,6 @@
#define IMX_DEFAULT_DMABUF_SIZE (64 * 1024)
-static inline void
-imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
- int dma, enum sdma_peripheral_type peripheral_type)
-{
- dma_data->dma_request = dma;
- dma_data->priority = DMA_PRIO_HIGH;
- dma_data->peripheral_type = peripheral_type;
-}
-
struct imx_pcm_fiq_params {
int irq;
void __iomem *base;
diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c
index 2e117311e582..4d99f4858a14 100644
--- a/sound/soc/fsl/imx-rpmsg.c
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -19,6 +19,7 @@
struct imx_rpmsg {
struct snd_soc_dai_link dai;
struct snd_soc_card card;
+ unsigned long sysclk;
};
static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = {
@@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Main MIC", NULL),
};
+static int imx_rpmsg_late_probe(struct snd_soc_card *card)
+{
+ struct imx_rpmsg *data = snd_soc_card_get_drvdata(card);
+ struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list,
+ struct snd_soc_pcm_runtime, list);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct device *dev = card->dev;
+ int ret;
+
+ if (!data->sysclk)
+ return 0;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dev, "failed to set sysclk in %s\n", __func__);
+ return ret;
+ }
+
+ return 0;
+}
+
static int imx_rpmsg_probe(struct platform_device *pdev)
{
struct snd_soc_dai_link_component *dlc;
@@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
data->dai.codecs->dai_name = "snd-soc-dummy-dai";
data->dai.codecs->name = "snd-soc-dummy";
} else {
+ struct clk *clk;
+
data->dai.codecs->of_node = args.np;
ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name);
if (ret) {
dev_err(&pdev->dev, "Unable to get codec_dai_name\n");
goto fail;
}
+
+ clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL);
+ if (!IS_ERR(clk))
+ data->sysclk = clk_get_rate(clk);
}
data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev);
@@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
data->card.owner = THIS_MODULE;
data->card.dapm_widgets = imx_rpmsg_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets);
+ data->card.late_probe = imx_rpmsg_late_probe;
/*
* Inoder to use common api to get card name and audio routing.
* Use parent of_node for this device, revert it after finishing using