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-rw-r--r--sound/soc/ti/Kconfig9
-rw-r--r--sound/soc/ti/Makefile2
-rw-r--r--sound/soc/ti/ams-delta.c9
-rw-r--r--sound/soc/ti/davinci-evm.c6
-rw-r--r--sound/soc/ti/davinci-mcasp.c3
-rw-r--r--sound/soc/ti/davinci-vcif.c4
-rw-r--r--sound/soc/ti/j721e-evm.c896
-rw-r--r--sound/soc/ti/n810.c4
-rw-r--r--sound/soc/ti/omap-abe-twl6040.c4
-rw-r--r--sound/soc/ti/omap-hdmi.c2
-rw-r--r--sound/soc/ti/omap-mcbsp-st.c3
-rw-r--r--sound/soc/ti/omap-mcbsp.c4
-rw-r--r--sound/soc/ti/omap-twl4030.c4
-rw-r--r--sound/soc/ti/omap3pandora.c2
-rw-r--r--sound/soc/ti/osk5912.c2
-rw-r--r--sound/soc/ti/rx51.c4
-rw-r--r--sound/soc/ti/sdma-pcm.c2
-rw-r--r--sound/soc/ti/sdma-pcm.h2
-rw-r--r--sound/soc/ti/udma-pcm.c2
-rw-r--r--sound/soc/ti/udma-pcm.h2
20 files changed, 937 insertions, 29 deletions
diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index c5408c129f34..1e6ab87e4460 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -219,5 +219,14 @@ config SND_SOC_DM365_VOICE_CODEC_MODULE
The is an internal symbol needed to ensure that the codec
and MFD driver can be built as loadable modules if necessary.
+config SND_SOC_J721E_EVM
+ tristate "SoC Audio support for j721e EVM"
+ depends on ARCH_K3_J721E_SOC || COMPILE_TEST
+ depends on I2C
+ select SND_SOC_PCM3168A_I2C
+ select SND_SOC_DAVINCI_MCASP
+ help
+ Say Y if you want to add support for SoC audio on j721e Common
+ Processor Board and Infotainment expansion board.
endmenu
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
index ea48c6679cc7..a21e5b0061de 100644
--- a/sound/soc/ti/Makefile
+++ b/sound/soc/ti/Makefile
@@ -34,6 +34,7 @@ snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
snd-soc-ams-delta-objs := ams-delta.o
snd-soc-omap-hdmi-objs := omap-hdmi.o
snd-soc-osk5912-objs := osk5912.o
+snd-soc-j721e-evm-objs := j721e-evm.o
obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o
obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o
@@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
obj-$(CONFIG_SND_SOC_OMAP_AMS_DELTA) += snd-soc-ams-delta.o
obj-$(CONFIG_SND_SOC_OMAP_HDMI) += snd-soc-omap-hdmi.o
obj-$(CONFIG_SND_SOC_OMAP_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_SOC_J721E_EVM) += snd-soc-j721e-evm.o
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index e17cd5e939f0..5c47de96c529 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -420,7 +420,7 @@ static struct snd_soc_ops ams_delta_ops;
* Shares hardware with codec config pulse generation */
static bool ams_delta_muted = 1;
-static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+static int ams_delta_mute(struct snd_soc_dai *dai, int mute, int direction)
{
int apply;
@@ -439,18 +439,19 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
/* Our codec DAI probably doesn't have its own .ops structure */
static const struct snd_soc_dai_ops ams_delta_dai_ops = {
- .digital_mute = ams_delta_digital_mute,
+ .mute_stream = ams_delta_mute,
+ .no_capture_mute = 1,
};
/* Will be used if the codec ever has its own digital_mute function */
static int ams_delta_startup(struct snd_pcm_substream *substream)
{
- return ams_delta_digital_mute(NULL, 0);
+ return ams_delta_digital_mute(NULL, 0, substream->stream);
}
static void ams_delta_shutdown(struct snd_pcm_substream *substream)
{
- ams_delta_digital_mute(NULL, 1);
+ ams_delta_digital_mute(NULL, 1, substream->stream);
}
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index 2cfbeebdfb41..105e56ab9cdc 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -28,7 +28,7 @@ struct snd_soc_card_drvdata_davinci {
static int evm_startup(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_card *soc_card = rtd->card;
struct snd_soc_card_drvdata_davinci *drvdata =
snd_soc_card_get_drvdata(soc_card);
@@ -41,7 +41,7 @@ static int evm_startup(struct snd_pcm_substream *substream)
static void evm_shutdown(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_card *soc_card = rtd->card;
struct snd_soc_card_drvdata_davinci *drvdata =
snd_soc_card_get_drvdata(soc_card);
@@ -53,7 +53,7 @@ static void evm_shutdown(struct snd_pcm_substream *substream)
static int evm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_soc_card *soc_card = rtd->card;
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index b93c1ee302c0..617440767c45 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1623,12 +1623,14 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.name = "davinci-mcasp.0",
.probe = davinci_mcasp_dai_probe,
.playback = {
+ .stream_name = "IIS Playback",
.channels_min = 1,
.channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
+ .stream_name = "IIS Capture",
.channels_min = 1,
.channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
@@ -1642,6 +1644,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.name = "davinci-mcasp.1",
.probe = davinci_mcasp_dai_probe,
.playback = {
+ .stream_name = "DIT Playback",
.channels_min = 1,
.channels_max = 384,
.rates = DAVINCI_MCASP_RATES,
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
index ee4d3ef821a1..f810123cc407 100644
--- a/sound/soc/ti/davinci-vcif.c
+++ b/sound/soc/ti/davinci-vcif.c
@@ -41,7 +41,7 @@ struct davinci_vcif_dev {
static void davinci_vcif_start(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct davinci_vcif_dev *davinci_vcif_dev =
snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
@@ -60,7 +60,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream)
static void davinci_vcif_stop(struct snd_pcm_substream *substream)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct davinci_vcif_dev *davinci_vcif_dev =
snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
new file mode 100644
index 000000000000..cb074af47a7d
--- /dev/null
+++ b/sound/soc/ti/j721e-evm.c
@@ -0,0 +1,896 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "davinci-mcasp.h"
+
+/*
+ * Maximum number of configuration entries for prefixes:
+ * CPB: 2 (mcasp10 + codec)
+ * IVI: 3 (mcasp0 + 2x codec)
+ */
+#define J721E_CODEC_CONF_COUNT 5
+
+#define J721E_AUDIO_DOMAIN_CPB 0
+#define J721E_AUDIO_DOMAIN_IVI 1
+
+#define J721E_CLK_PARENT_48000 0
+#define J721E_CLK_PARENT_44100 1
+
+#define J721E_MAX_CLK_HSDIV 128
+#define PCM1368A_MAX_SYSCLK 36864000
+
+#define J721E_DAI_FMT (SND_SOC_DAIFMT_RIGHT_J | \
+ SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBS_CFS)
+
+enum j721e_board_type {
+ J721E_BOARD_CPB = 1,
+ J721E_BOARD_CPB_IVI,
+};
+
+struct j721e_audio_match_data {
+ enum j721e_board_type board_type;
+ int num_links;
+ unsigned int pll_rates[2];
+};
+
+static unsigned int ratios_for_pcm3168a[] = {
+ 256,
+ 512,
+ 768,
+};
+
+struct j721e_audio_clocks {
+ struct clk *target;
+ struct clk *parent[2];
+};
+
+struct j721e_audio_domain {
+ struct j721e_audio_clocks codec;
+ struct j721e_audio_clocks mcasp;
+ int parent_clk_id;
+
+ int active;
+ unsigned int active_link;
+ unsigned int rate;
+};
+
+struct j721e_priv {
+ struct device *dev;
+ struct snd_soc_card card;
+ struct snd_soc_dai_link *dai_links;
+ struct snd_soc_codec_conf codec_conf[J721E_CODEC_CONF_COUNT];
+ struct snd_interval rate_range;
+ const struct j721e_audio_match_data *match_data;
+ u32 pll_rates[2];
+ unsigned int hsdiv_rates[2];
+
+ struct j721e_audio_domain audio_domains[2];
+
+ struct mutex mutex;
+};
+
+static const struct snd_soc_dapm_widget j721e_cpb_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("CPB Stereo HP 1", NULL),
+ SND_SOC_DAPM_HP("CPB Stereo HP 2", NULL),
+ SND_SOC_DAPM_HP("CPB Stereo HP 3", NULL),
+ SND_SOC_DAPM_LINE("CPB Line Out", NULL),
+ SND_SOC_DAPM_MIC("CPB Stereo Mic 1", NULL),
+ SND_SOC_DAPM_MIC("CPB Stereo Mic 2", NULL),
+ SND_SOC_DAPM_LINE("CPB Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_cpb_dapm_routes[] = {
+ {"CPB Stereo HP 1", NULL, "codec-1 AOUT1L"},
+ {"CPB Stereo HP 1", NULL, "codec-1 AOUT1R"},
+ {"CPB Stereo HP 2", NULL, "codec-1 AOUT2L"},
+ {"CPB Stereo HP 2", NULL, "codec-1 AOUT2R"},
+ {"CPB Stereo HP 3", NULL, "codec-1 AOUT3L"},
+ {"CPB Stereo HP 3", NULL, "codec-1 AOUT3R"},
+ {"CPB Line Out", NULL, "codec-1 AOUT4L"},
+ {"CPB Line Out", NULL, "codec-1 AOUT4R"},
+
+ {"codec-1 AIN1L", NULL, "CPB Stereo Mic 1"},
+ {"codec-1 AIN1R", NULL, "CPB Stereo Mic 1"},
+ {"codec-1 AIN2L", NULL, "CPB Stereo Mic 2"},
+ {"codec-1 AIN2R", NULL, "CPB Stereo Mic 2"},
+ {"codec-1 AIN3L", NULL, "CPB Line In"},
+ {"codec-1 AIN3R", NULL, "CPB Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_a_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("IVI A Line Out 1", NULL),
+ SND_SOC_DAPM_LINE("IVI A Line Out 2", NULL),
+ SND_SOC_DAPM_LINE("IVI A Line Out 3", NULL),
+ SND_SOC_DAPM_LINE("IVI A Line Out 4", NULL),
+ SND_SOC_DAPM_MIC("IVI A Stereo Mic 1", NULL),
+ SND_SOC_DAPM_MIC("IVI A Stereo Mic 2", NULL),
+ SND_SOC_DAPM_LINE("IVI A Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_a_dapm_routes[] = {
+ {"IVI A Line Out 1", NULL, "codec-a AOUT1L"},
+ {"IVI A Line Out 1", NULL, "codec-a AOUT1R"},
+ {"IVI A Line Out 2", NULL, "codec-a AOUT2L"},
+ {"IVI A Line Out 2", NULL, "codec-a AOUT2R"},
+ {"IVI A Line Out 3", NULL, "codec-a AOUT3L"},
+ {"IVI A Line Out 3", NULL, "codec-a AOUT3R"},
+ {"IVI A Line Out 4", NULL, "codec-a AOUT4L"},
+ {"IVI A Line Out 4", NULL, "codec-a AOUT4R"},
+
+ {"codec-a AIN1L", NULL, "IVI A Stereo Mic 1"},
+ {"codec-a AIN1R", NULL, "IVI A Stereo Mic 1"},
+ {"codec-a AIN2L", NULL, "IVI A Stereo Mic 2"},
+ {"codec-a AIN2R", NULL, "IVI A Stereo Mic 2"},
+ {"codec-a AIN3L", NULL, "IVI A Line In"},
+ {"codec-a AIN3R", NULL, "IVI A Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_b_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("IVI B Line Out 1", NULL),
+ SND_SOC_DAPM_LINE("IVI B Line Out 2", NULL),
+ SND_SOC_DAPM_LINE("IVI B Line Out 3", NULL),
+ SND_SOC_DAPM_LINE("IVI B Line Out 4", NULL),
+ SND_SOC_DAPM_MIC("IVI B Stereo Mic 1", NULL),
+ SND_SOC_DAPM_MIC("IVI B Stereo Mic 2", NULL),
+ SND_SOC_DAPM_LINE("IVI B Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_b_dapm_routes[] = {
+ {"IVI B Line Out 1", NULL, "codec-b AOUT1L"},
+ {"IVI B Line Out 1", NULL, "codec-b AOUT1R"},
+ {"IVI B Line Out 2", NULL, "codec-b AOUT2L"},
+ {"IVI B Line Out 2", NULL, "codec-b AOUT2R"},
+ {"IVI B Line Out 3", NULL, "codec-b AOUT3L"},
+ {"IVI B Line Out 3", NULL, "codec-b AOUT3R"},
+ {"IVI B Line Out 4", NULL, "codec-b AOUT4L"},
+ {"IVI B Line Out 4", NULL, "codec-b AOUT4R"},
+
+ {"codec-b AIN1L", NULL, "IVI B Stereo Mic 1"},
+ {"codec-b AIN1R", NULL, "IVI B Stereo Mic 1"},
+ {"codec-b AIN2L", NULL, "IVI B Stereo Mic 2"},
+ {"codec-b AIN2R", NULL, "IVI B Stereo Mic 2"},
+ {"codec-b AIN3L", NULL, "IVI B Line In"},
+ {"codec-b AIN3R", NULL, "IVI B Line In"},
+};
+
+static int j721e_configure_refclk(struct j721e_priv *priv,
+ unsigned int audio_domain, unsigned int rate)
+{
+ struct j721e_audio_domain *domain = &priv->audio_domains[audio_domain];
+ unsigned int scki;
+ int ret = -EINVAL;
+ int i, clk_id;
+
+ if (!(rate % 8000) && priv->pll_rates[J721E_CLK_PARENT_48000])
+ clk_id = J721E_CLK_PARENT_48000;
+ else if (!(rate % 11025) && priv->pll_rates[J721E_CLK_PARENT_44100])
+ clk_id = J721E_CLK_PARENT_44100;
+ else
+ return ret;
+
+ for (i = 0; i < ARRAY_SIZE(ratios_for_pcm3168a); i++) {
+ scki = ratios_for_pcm3168a[i] * rate;
+
+ if (priv->pll_rates[clk_id] / scki <= J721E_MAX_CLK_HSDIV) {
+ ret = 0;
+ break;
+ }
+ }
+
+ if (ret) {
+ dev_err(priv->dev, "No valid clock configuration for %u Hz\n",
+ rate);
+ return ret;
+ }
+
+ if (priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+ dev_dbg(priv->dev,
+ "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
+ audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI",
+ rate,
+ clk_id == J721E_CLK_PARENT_48000 ? "PLL4" : "PLL15",
+ ratios_for_pcm3168a[i], scki);
+
+ if (domain->parent_clk_id != clk_id) {
+ ret = clk_set_parent(domain->codec.target,
+ domain->codec.parent[clk_id]);
+ if (ret)
+ return ret;
+
+ ret = clk_set_parent(domain->mcasp.target,
+ domain->mcasp.parent[clk_id]);
+ if (ret)
+ return ret;
+
+ domain->parent_clk_id = clk_id;
+ }
+
+ ret = clk_set_rate(domain->codec.target, scki);
+ if (ret) {
+ dev_err(priv->dev, "codec set rate failed for %u Hz\n",
+ scki);
+ return ret;
+ }
+
+ ret = clk_set_rate(domain->mcasp.target, scki);
+ if (!ret) {
+ priv->hsdiv_rates[domain->parent_clk_id] = scki;
+ } else {
+ dev_err(priv->dev, "mcasp set rate failed for %u Hz\n",
+ scki);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static int j721e_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *t = rule->private;
+
+ return snd_interval_refine(hw_param_interval(params, rule->var), t);
+}
+
+static int j721e_audio_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ unsigned int domain_id = rtd->dai_link->id;
+ struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ unsigned int active_rate;
+ int ret = 0;
+ int i;
+
+ mutex_lock(&priv->mutex);
+
+ domain->active++;
+
+ if (priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].rate)
+ active_rate = priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].rate;
+ else
+ active_rate = priv->audio_domains[J721E_AUDIO_DOMAIN_IVI].rate;
+
+ if (active_rate)
+ ret = snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ active_rate);
+ else
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ j721e_rule_rate, &priv->rate_range,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
+ mutex_unlock(&priv->mutex);
+
+ if (ret)
+ return ret;
+
+ /* Reset TDM slots to 32 */
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_card *card = rtd->card;
+ struct j721e_priv *priv = snd_soc_card_get_drvdata(card);
+ unsigned int domain_id = rtd->dai_link->id;
+ struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ unsigned int sysclk_rate;
+ int slot_width = 32;
+ int ret;
+ int i;
+
+ mutex_lock(&priv->mutex);
+
+ if (domain->rate && domain->rate != params_rate(params)) {
+ ret = -EINVAL;
+ goto out;
+ }
+
+ if (params_width(params) == 16)
+ slot_width = 16;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, slot_width);
+ if (ret && ret != -ENOTSUPP)
+ goto out;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2,
+ slot_width);
+ if (ret && ret != -ENOTSUPP)
+ goto out;
+ }
+
+ ret = j721e_configure_refclk(priv, domain_id, params_rate(params));
+ if (ret)
+ goto out;
+
+ sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+ SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(priv->dev,
+ "codec set_sysclk failed for %u Hz\n",
+ sysclk_rate);
+ goto out;
+ }
+ }
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+ sysclk_rate, SND_SOC_CLOCK_IN);
+
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(priv->dev, "mcasp set_sysclk failed for %u Hz\n",
+ sysclk_rate);
+ } else {
+ domain->rate = params_rate(params);
+ ret = 0;
+ }
+
+out:
+ mutex_unlock(&priv->mutex);
+ return ret;
+}
+
+static void j721e_audio_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ unsigned int domain_id = rtd->dai_link->id;
+ struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+
+ mutex_lock(&priv->mutex);
+
+ domain->active--;
+ if (!domain->active) {
+ domain->rate = 0;
+ domain->active_link = 0;
+ }
+
+ mutex_unlock(&priv->mutex);
+}
+
+static const struct snd_soc_ops j721e_audio_ops = {
+ .startup = j721e_audio_startup,
+ .hw_params = j721e_audio_hw_params,
+ .shutdown = j721e_audio_shutdown,
+};
+
+static int j721e_audio_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ unsigned int domain_id = rtd->dai_link->id;
+ struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ unsigned int sysclk_rate;
+ int i, ret;
+
+ /* Set up initial clock configuration */
+ ret = j721e_configure_refclk(priv, domain_id, 48000);
+ if (ret)
+ return ret;
+
+ sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+ SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+ sysclk_rate, SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+
+ /* Set initial tdm slots */
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int j721e_audio_init_ivi(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
+
+ snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_a_dapm_widgets,
+ ARRAY_SIZE(j721e_ivi_codec_a_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, j721e_codec_a_dapm_routes,
+ ARRAY_SIZE(j721e_codec_a_dapm_routes));
+ snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_b_dapm_widgets,
+ ARRAY_SIZE(j721e_ivi_codec_b_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, j721e_codec_b_dapm_routes,
+ ARRAY_SIZE(j721e_codec_b_dapm_routes));
+
+ return j721e_audio_init(rtd);
+}
+
+static int j721e_get_clocks(struct device *dev,
+ struct j721e_audio_clocks *clocks, char *prefix)
+{
+ struct clk *parent;
+ char *clk_name;
+ int ret;
+
+ clocks->target = devm_clk_get(dev, prefix);
+ if (IS_ERR(clocks->target)) {
+ ret = PTR_ERR(clocks->target);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "failed to acquire %s: %d\n",
+ prefix, ret);
+ return ret;
+ }
+
+ clk_name = kasprintf(GFP_KERNEL, "%s-48000", prefix);
+ if (clk_name) {
+ parent = devm_clk_get(dev, clk_name);
+ kfree(clk_name);
+ if (IS_ERR(parent)) {
+ ret = PTR_ERR(parent);
+ if (ret == -EPROBE_DEFER)
+ return ret;
+
+ dev_dbg(dev, "no 48KHz parent for %s: %d\n", prefix, ret);
+ parent = NULL;
+ }
+ clocks->parent[J721E_CLK_PARENT_48000] = parent;
+ } else {
+ return -ENOMEM;
+ }
+
+ clk_name = kasprintf(GFP_KERNEL, "%s-44100", prefix);
+ if (clk_name) {
+ parent = devm_clk_get(dev, clk_name);
+ kfree(clk_name);
+ if (IS_ERR(parent)) {
+ ret = PTR_ERR(parent);
+ if (ret == -EPROBE_DEFER)
+ return ret;
+
+ dev_dbg(dev, "no 44.1KHz parent for %s: %d\n", prefix, ret);
+ parent = NULL;
+ }
+ clocks->parent[J721E_CLK_PARENT_44100] = parent;
+ } else {
+ return -ENOMEM;
+ }
+
+ if (!clocks->parent[J721E_CLK_PARENT_44100] &&
+ !clocks->parent[J721E_CLK_PARENT_48000]) {
+ dev_err(dev, "At least one parent clock is needed for %s\n",
+ prefix);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct j721e_audio_match_data j721e_cpb_data = {
+ .board_type = J721E_BOARD_CPB,
+ .num_links = 2, /* CPB pcm3168a */
+ .pll_rates = {
+ [J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+ [J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+ },
+};
+
+static const struct j721e_audio_match_data j721e_cpb_ivi_data = {
+ .board_type = J721E_BOARD_CPB_IVI,
+ .num_links = 4, /* CPB pcm3168a + 2x pcm3168a on IVI */
+ .pll_rates = {
+ [J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+ [J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+ },
+};
+
+static const struct of_device_id j721e_audio_of_match[] = {
+ {
+ .compatible = "ti,j721e-cpb-audio",
+ .data = &j721e_cpb_data,
+ }, {
+ .compatible = "ti,j721e-cpb-ivi-audio",
+ .data = &j721e_cpb_ivi_data,
+ },
+ { },
+};
+MODULE_DEVICE_TABLE(of, j721e_audio_of_match);
+
+static int j721e_calculate_rate_range(struct j721e_priv *priv)
+{
+ const struct j721e_audio_match_data *match_data = priv->match_data;
+ struct j721e_audio_clocks *domain_clocks;
+ unsigned int min_rate, max_rate, pll_rate;
+ struct clk *pll;
+
+ domain_clocks = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].mcasp;
+
+ pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_44100]);
+ if (IS_ERR_OR_NULL(pll)) {
+ priv->pll_rates[J721E_CLK_PARENT_44100] =
+ match_data->pll_rates[J721E_CLK_PARENT_44100];
+ } else {
+ priv->pll_rates[J721E_CLK_PARENT_44100] = clk_get_rate(pll);
+ clk_put(pll);
+ }
+
+ pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_48000]);
+ if (IS_ERR_OR_NULL(pll)) {
+ priv->pll_rates[J721E_CLK_PARENT_48000] =
+ match_data->pll_rates[J721E_CLK_PARENT_48000];
+ } else {
+ priv->pll_rates[J721E_CLK_PARENT_48000] = clk_get_rate(pll);
+ clk_put(pll);
+ }
+
+ if (!priv->pll_rates[J721E_CLK_PARENT_44100] &&
+ !priv->pll_rates[J721E_CLK_PARENT_48000]) {
+ dev_err(priv->dev, "At least one PLL is needed\n");
+ return -EINVAL;
+ }
+
+ if (priv->pll_rates[J721E_CLK_PARENT_44100])
+ pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+ else
+ pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+
+ min_rate = pll_rate / J721E_MAX_CLK_HSDIV;
+ min_rate /= ratios_for_pcm3168a[ARRAY_SIZE(ratios_for_pcm3168a) - 1];
+
+ if (priv->pll_rates[J721E_CLK_PARENT_48000])
+ pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+ else
+ pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+
+ if (pll_rate > PCM1368A_MAX_SYSCLK)
+ pll_rate = PCM1368A_MAX_SYSCLK;
+
+ max_rate = pll_rate / ratios_for_pcm3168a[0];
+
+ snd_interval_any(&priv->rate_range);
+ priv->rate_range.min = min_rate;
+ priv->rate_range.max = max_rate;
+
+ return 0;
+}
+
+static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
+ int *conf_idx)
+{
+ struct device_node *node = priv->dev->of_node;
+ struct snd_soc_dai_link_component *compnent;
+ struct device_node *dai_node, *codec_node;
+ struct j721e_audio_domain *domain;
+ int comp_count, comp_idx;
+ int ret;
+
+ dai_node = of_parse_phandle(node, "ti,cpb-mcasp", 0);
+ if (!dai_node) {
+ dev_err(priv->dev, "CPB McASP node is not provided\n");
+ return -EINVAL;
+ }
+
+ codec_node = of_parse_phandle(node, "ti,cpb-codec", 0);
+ if (!codec_node) {
+ dev_err(priv->dev, "CPB codec node is not provided\n");
+ return -EINVAL;
+ }
+
+ domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB];
+ ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki");
+ if (ret)
+ return ret;
+
+ ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk");
+ if (ret)
+ return ret;
+
+ /*
+ * Common Processor Board, two links
+ * Link 1: McASP10 -> pcm3168a_1 DAC
+ * Link 2: McASP10 <- pcm3168a_1 ADC
+ */
+ comp_count = 6;
+ compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+ GFP_KERNEL);
+ if (!compnent)
+ return -ENOMEM;
+
+ comp_idx = 0;
+ priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_cpus = 1;
+ priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_codecs = 1;
+ priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_platforms = 1;
+
+ priv->dai_links[*link_idx].name = "CPB PCM3168A Playback";
+ priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+ priv->dai_links[*link_idx].cpus->of_node = dai_node;
+ priv->dai_links[*link_idx].platforms->of_node = dai_node;
+ priv->dai_links[*link_idx].codecs->of_node = codec_node;
+ priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-dac";
+ priv->dai_links[*link_idx].playback_only = 1;
+ priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+ priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+ priv->dai_links[*link_idx].init = j721e_audio_init;
+ priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+ (*link_idx)++;
+
+ priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_cpus = 1;
+ priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_codecs = 1;
+ priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_platforms = 1;
+
+ priv->dai_links[*link_idx].name = "CPB PCM3168A Capture";
+ priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+ priv->dai_links[*link_idx].cpus->of_node = dai_node;
+ priv->dai_links[*link_idx].platforms->of_node = dai_node;
+ priv->dai_links[*link_idx].codecs->of_node = codec_node;
+ priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-adc";
+ priv->dai_links[*link_idx].capture_only = 1;
+ priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+ priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+ priv->dai_links[*link_idx].init = j721e_audio_init;
+ priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+ (*link_idx)++;
+
+ priv->codec_conf[*conf_idx].dlc.of_node = codec_node;
+ priv->codec_conf[*conf_idx].name_prefix = "codec-1";
+ (*conf_idx)++;
+ priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+ priv->codec_conf[*conf_idx].name_prefix = "McASP10";
+ (*conf_idx)++;
+
+ return 0;
+}
+
+static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
+ int *conf_idx)
+{
+ struct device_node *node = priv->dev->of_node;
+ struct snd_soc_dai_link_component *compnent;
+ struct device_node *dai_node, *codeca_node, *codecb_node;
+ struct j721e_audio_domain *domain;
+ int comp_count, comp_idx;
+ int ret;
+
+ if (priv->match_data->board_type != J721E_BOARD_CPB_IVI)
+ return 0;
+
+ dai_node = of_parse_phandle(node, "ti,ivi-mcasp", 0);
+ if (!dai_node) {
+ dev_err(priv->dev, "IVI McASP node is not provided\n");
+ return -EINVAL;
+ }
+
+ codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0);
+ if (!codeca_node) {
+ dev_err(priv->dev, "IVI codec-a node is not provided\n");
+ return -EINVAL;
+ }
+
+ codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0);
+ if (!codecb_node) {
+ dev_warn(priv->dev, "IVI codec-b node is not provided\n");
+ return 0;
+ }
+
+ domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI];
+ ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki");
+ if (ret)
+ return ret;
+
+ ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk");
+ if (ret)
+ return ret;
+
+ /*
+ * IVI extension, two links
+ * Link 1: McASP0 -> pcm3168a_a DAC
+ * \> pcm3168a_b DAC
+ * Link 2: McASP0 <- pcm3168a_a ADC
+ * \ pcm3168a_b ADC
+ */
+ comp_count = 8;
+ compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+ GFP_KERNEL);
+ if (!compnent)
+ return -ENOMEM;
+
+ comp_idx = 0;
+ priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_cpus = 1;
+ priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_platforms = 1;
+ priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+ priv->dai_links[*link_idx].num_codecs = 2;
+ comp_idx += 2;
+
+ priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Playback";
+ priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+ priv->dai_links[*link_idx].cpus->of_node = dai_node;
+ priv->dai_links[*link_idx].platforms->of_node = dai_node;
+ priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+ priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-dac";
+ priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+ priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-dac";
+ priv->dai_links[*link_idx].playback_only = 1;
+ priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+ priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+ priv->dai_links[*link_idx].init = j721e_audio_init_ivi;
+ priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+ (*link_idx)++;
+
+ priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_cpus = 1;
+ priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+ priv->dai_links[*link_idx].num_platforms = 1;
+ priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+ priv->dai_links[*link_idx].num_codecs = 2;
+
+ priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Capture";
+ priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+ priv->dai_links[*link_idx].cpus->of_node = dai_node;
+ priv->dai_links[*link_idx].platforms->of_node = dai_node;
+ priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+ priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-adc";
+ priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+ priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-adc";
+ priv->dai_links[*link_idx].capture_only = 1;
+ priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+ priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+ priv->dai_links[*link_idx].init = j721e_audio_init;
+ priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+ (*link_idx)++;
+
+ priv->codec_conf[*conf_idx].dlc.of_node = codeca_node;
+ priv->codec_conf[*conf_idx].name_prefix = "codec-a";
+ (*conf_idx)++;
+
+ priv->codec_conf[*conf_idx].dlc.of_node = codecb_node;
+ priv->codec_conf[*conf_idx].name_prefix = "codec-b";
+ (*conf_idx)++;
+
+ priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+ priv->codec_conf[*conf_idx].name_prefix = "McASP0";
+ (*conf_idx)++;
+
+ return 0;
+}
+
+static int j721e_soc_probe(struct platform_device *pdev)
+{
+ struct device_node *node = pdev->dev.of_node;
+ struct snd_soc_card *card;
+ const struct of_device_id *match;
+ struct j721e_priv *priv;
+ int link_cnt, conf_cnt, ret;
+
+ if (!node) {
+ dev_err(&pdev->dev, "of node is missing.\n");
+ return -ENODEV;
+ }
+
+ match = of_match_node(j721e_audio_of_match, node);
+ if (!match) {
+ dev_err(&pdev->dev, "No compatible match found\n");
+ return -ENODEV;
+ }
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->match_data = match->data;
+
+ priv->dai_links = devm_kcalloc(&pdev->dev, priv->match_data->num_links,
+ sizeof(*priv->dai_links), GFP_KERNEL);
+ if (!priv->dai_links)
+ return -ENOMEM;
+
+ priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].parent_clk_id = -1;
+ priv->audio_domains[J721E_AUDIO_DOMAIN_IVI].parent_clk_id = -1;
+ priv->dev = &pdev->dev;
+ card = &priv->card;
+ card->dev = &pdev->dev;
+ card->owner = THIS_MODULE;
+ card->dapm_widgets = j721e_cpb_dapm_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(j721e_cpb_dapm_widgets);
+ card->dapm_routes = j721e_cpb_dapm_routes;
+ card->num_dapm_routes = ARRAY_SIZE(j721e_cpb_dapm_routes);
+ card->fully_routed = 1;
+
+ if (snd_soc_of_parse_card_name(card, "model")) {
+ dev_err(&pdev->dev, "Card name is not provided\n");
+ return -ENODEV;
+ }
+
+ link_cnt = 0;
+ conf_cnt = 0;
+ ret = j721e_soc_probe_cpb(priv, &link_cnt, &conf_cnt);
+ if (ret)
+ return ret;
+
+ ret = j721e_soc_probe_ivi(priv, &link_cnt, &conf_cnt);
+ if (ret)
+ return ret;
+
+ card->dai_link = priv->dai_links;
+ card->num_links = link_cnt;
+
+ card->codec_conf = priv->codec_conf;
+ card->num_configs = conf_cnt;
+
+ ret = j721e_calculate_rate_range(priv);
+ if (ret)
+ return ret;
+
+ snd_soc_card_set_drvdata(card, priv);
+
+ mutex_init(&priv->mutex);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
+ if (ret)
+ dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+static struct platform_driver j721e_soc_driver = {
+ .driver = {
+ .name = "j721e-audio",
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = of_match_ptr(j721e_audio_of_match),
+ },
+ .probe = j721e_soc_probe,
+};
+
+module_platform_driver(j721e_soc_driver);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("ASoC machine driver for j721e Common Processor Board");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index a1672b479cb7..2802a33b9c5f 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -84,7 +84,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
static int n810_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
@@ -100,7 +100,7 @@ static void n810_shutdown(struct snd_pcm_substream *substream)
static int n810_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int err;
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
index 61e45fea5dd8..16ea039ff865 100644
--- a/sound/soc/ti/omap-abe-twl6040.c
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -45,7 +45,7 @@ static struct platform_device *dmic_codec_dev;
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
@@ -77,7 +77,7 @@ static const struct snd_soc_ops omap_abe_ops = {
static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret = 0;
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index def2a0ce8886..3328c02f93c7 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/sound/soc/ti/omap-hdmi.c
@@ -2,7 +2,7 @@
/*
* omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
*
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
*
* Author: Jyri Sarha <jsarha@ti.com>
*/
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 5a32b54bbf3b..0bc7d26c660a 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -142,11 +142,8 @@ static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir)
static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp)
{
- u16 w;
struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- w = MCBSP_ST_READ(mcbsp, SSELCR);
-
MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) |
ST_CH1GAIN(st_data->ch1gain));
}
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 32e3ccdbb7a2..6025b30bbe77 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -731,7 +731,7 @@ err_st:
static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
unsigned int packet_size)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
int words;
@@ -896,7 +896,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay(
struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
u16 fifo_use;
diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c
index 92dbe2c67290..1da05a6cdc9f 100644
--- a/sound/soc/ti/omap-twl4030.c
+++ b/sound/soc/ti/omap-twl4030.c
@@ -2,7 +2,7 @@
/*
* omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec
*
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2012 Texas Instruments Incorporated - https://www.ti.com
* All rights reserved.
*
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
@@ -38,7 +38,7 @@ struct omap_twl4030 {
static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
unsigned int fmt;
switch (params_channels(params)) {
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
index b04146311b31..a287e9747c2a 100644
--- a/sound/soc/ti/omap3pandora.c
+++ b/sound/soc/ti/omap3pandora.c
@@ -31,7 +31,7 @@ static struct regulator *omap3pandora_dac_reg;
static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
index e01485cc51a1..40e29dda7e7a 100644
--- a/sound/soc/ti/osk5912.c
+++ b/sound/soc/ti/osk5912.c
@@ -38,7 +38,7 @@ static void osk_shutdown(struct snd_pcm_substream *substream)
static int osk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int err;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2a714a004163..2176a95201bf 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -90,7 +90,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
static int rx51_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
@@ -102,7 +102,7 @@ static int rx51_startup(struct snd_pcm_substream *substream)
static int rx51_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c
index 2b0bc234e1b6..9e7691103f05 100644
--- a/sound/soc/ti/sdma-pcm.c
+++ b/sound/soc/ti/sdma-pcm.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*/
diff --git a/sound/soc/ti/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h
index cb0627c8dd34..c19efb4c043d 100644
--- a/sound/soc/ti/sdma-pcm.h
+++ b/sound/soc/ti/sdma-pcm.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0 */
/*
- * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*/
diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c
index 39830caaaf7c..2ff0f518aba5 100644
--- a/sound/soc/ti/udma-pcm.c
+++ b/sound/soc/ti/udma-pcm.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com
* Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*/
diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h
index 54111e7312c1..9ed588fd79b9 100644
--- a/sound/soc/ti/udma-pcm.h
+++ b/sound/soc/ti/udma-pcm.h
@@ -1,6 +1,6 @@
/* SPDX-License-Identifier: GPL-2.0 */
/*
- * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
*/
#ifndef __UDMA_PCM_H__
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