The patchset adds NXP Audio Mixer (AUDMIX) device and machine
drivers and related DT bindings documentation.
Changes since V4:
1. Removed "model" attribute from device driver DT bindings documentation
as suggested by Nicolin.
Changes since V3:
1. Removed machine driver DT bindings documentation.
2. Trigger machine driver probe from device driver as suggested by Nicolin.
Changes since V2:
1. Moved "dais" node from machine driver DTS node to device driver DTS node
as suggested by Rob.
Changes since V1:
1. Original patch split into distinct patches for the device driver and
DT binding documentation.
2. Replaced AMIX with AUDMIX in both code and file names as it looks more
RM-compliant.
3. Removed polarity control from CPU DAI driver as suggested by Nicolin.
4. Added machine driver and related DT binding documentation.
Viorel Suman (3):
ASoC: fsl: Add Audio Mixer CPU DAI driver
ASoC: add fsl_audmix DT binding documentation
ASoC: fsl: Add Audio Mixer machine driver
.../devicetree/bindings/sound/fsl,audmix.txt | 50 ++
sound/soc/fsl/Kconfig | 16 +
sound/soc/fsl/Makefile | 5 +
sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++
sound/soc/fsl/fsl_audmix.h | 102 ++++
sound/soc/fsl/imx-audmix.c | 327 ++++++++++++
6 files changed, 1078 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
create mode 100644 sound/soc/fsl/fsl_audmix.c
create mode 100644 sound/soc/fsl/fsl_audmix.h
create mode 100644 sound/soc/fsl/imx-audmix.c
--
2.7.4
Add the DT binding documentation for NXP Audio Mixer
CPU DAI driver.
Signed-off-by: Viorel Suman <[email protected]>
Acked-by: Nicolin Chen <[email protected]>
Acked-by: Rob Herring <[email protected]>
---
.../devicetree/bindings/sound/fsl,audmix.txt | 50 ++++++++++++++++++++++
1 file changed, 50 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
new file mode 100644
index 0000000..840b7e0
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
@@ -0,0 +1,50 @@
+NXP Audio Mixer (AUDMIX).
+
+The Audio Mixer is a on-chip functional module that allows mixing of two
+audio streams into a single audio stream. Audio Mixer has two input serial
+audio interfaces. These are driven by two Synchronous Audio interface
+modules (SAI). Each input serial interface carries 8 audio channels in its
+frame in TDM manner. Mixer mixes audio samples of corresponding channels
+from two interfaces into a single sample. Before mixing, audio samples of
+two inputs can be attenuated based on configuration. The output of the
+Audio Mixer is also a serial audio interface. Like input interfaces it has
+the same TDM frame format. This output is used to drive the serial DAC TDM
+interface of audio codec and also sent to the external pins along with the
+receive path of normal audio SAI module for readback by the CPU.
+
+The output of Audio Mixer can be selected from any of the three streams
+ - serial audio input 1
+ - serial audio input 2
+ - mixed audio
+
+Mixing operation is independent of audio sample rate but the two audio
+input streams must have same audio sample rate with same number of channels
+in TDM frame to be eligible for mixing.
+
+Device driver required properties:
+=================================
+ - compatible : Compatible list, contains "fsl,imx8qm-audmix"
+
+ - reg : Offset and length of the register set for the device.
+
+ - clocks : Must contain an entry for each entry in clock-names.
+
+ - clock-names : Must include the "ipg" for register access.
+
+ - power-domains : Must contain the phandle to AUDMIX power domain node
+
+ - dais : Must contain a list of phandles to AUDMIX connected
+ DAIs. The current implementation requires two phandles
+ to SAI interfaces to be provided, the first SAI in the
+ list being used to route the AUDMIX output.
+
+Device driver configuration example:
+======================================
+ audmix: audmix@59840000 {
+ compatible = "fsl,imx8qm-audmix";
+ reg = <0x0 0x59840000 0x0 0x10000>;
+ clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
+ clock-names = "ipg";
+ power-domains = <&pd_audmix>;
+ dais = <&sai4>, <&sai5>;
+ };
--
2.7.4
This patch implements Audio Mixer machine driver for NXP iMX8 SOCs.
It connects together Audio Mixer and related SAI instances.
Signed-off-by: Viorel Suman <[email protected]>
Acked-by: Nicolin Chen <[email protected]>
---
sound/soc/fsl/Kconfig | 9 ++
sound/soc/fsl/Makefile | 2 +
sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 338 insertions(+)
create mode 100644 sound/soc/fsl/imx-audmix.c
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 0af2e056..d87c842 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -303,6 +303,15 @@ config SND_SOC_FSL_ASOC_CARD
CS4271, CS4272 and SGTL5000.
Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+config SND_SOC_IMX_AUDMIX
+ tristate "SoC Audio support for i.MX boards with AUDMIX"
+ select SND_SOC_FSL_AUDMIX
+ select SND_SOC_FSL_SAI
+ help
+ SoC Audio support for i.MX boards with Audio Mixer
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ an Audio Mixer.
+
endif # SND_IMX_SOC
endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 4172d5a..c0dd044 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -62,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-spdif-objs := imx-spdif.o
snd-soc-imx-mc13783-objs := imx-mc13783.o
+snd-soc-imx-audmix-objs := imx-audmix.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
@@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
+obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
new file mode 100644
index 0000000..72e37ca
--- /dev/null
+++ b/sound/soc/fsl/imx-audmix.c
@@ -0,0 +1,327 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright 2017 NXP
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <linux/pm_runtime.h>
+#include "fsl_sai.h"
+#include "fsl_audmix.h"
+
+struct imx_audmix {
+ struct platform_device *pdev;
+ struct snd_soc_card card;
+ struct platform_device *audmix_pdev;
+ struct platform_device *out_pdev;
+ struct clk *cpu_mclk;
+ int num_dai;
+ struct snd_soc_dai_link *dai;
+ int num_dai_conf;
+ struct snd_soc_codec_conf *dai_conf;
+ int num_dapm_routes;
+ struct snd_soc_dapm_route *dapm_routes;
+};
+
+static const u32 imx_audmix_rates[] = {
+ 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000,
+};
+
+static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = {
+ .count = ARRAY_SIZE(imx_audmix_rates),
+ .list = imx_audmix_rates,
+};
+
+static int imx_audmix_fe_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct device *dev = rtd->card->dev;
+ unsigned long clk_rate = clk_get_rate(priv->cpu_mclk);
+ int ret;
+
+ if (clk_rate % 24576000 == 0) {
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &imx_audmix_rate_constraints);
+ if (ret < 0)
+ return ret;
+ } else {
+ dev_warn(dev, "mclk may be not supported %lu\n", clk_rate);
+ }
+
+ ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS,
+ 1, 8);
+ if (ret < 0)
+ return ret;
+
+ return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT,
+ FSL_AUDMIX_FORMATS);
+}
+
+static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->card->dev;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
+ u32 channels = params_channels(params);
+ int ret, dir;
+
+ /* For playback the AUDMIX is slave, and for record is master */
+ fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM;
+ dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN;
+
+ /* set DAI configuration */
+ ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+ if (ret) {
+ dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir);
+ if (ret) {
+ dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * Per datasheet, AUDMIX expects 8 slots and 32 bits
+ * for every slot in TDM mode.
+ */
+ ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1,
+ BIT(channels) - 1, 8, 32);
+ if (ret)
+ dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret);
+
+ return ret;
+}
+
+static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->card->dev;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
+ int ret;
+
+ if (!tx)
+ return 0;
+
+ /* For playback the AUDMIX is slave */
+ fmt |= SND_SOC_DAIFMT_CBM_CFM;
+
+ /* set AUDMIX DAI configuration */
+ ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+ if (ret)
+ dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret);
+
+ return ret;
+}
+
+static struct snd_soc_ops imx_audmix_fe_ops = {
+ .startup = imx_audmix_fe_startup,
+ .hw_params = imx_audmix_fe_hw_params,
+};
+
+static struct snd_soc_ops imx_audmix_be_ops = {
+ .hw_params = imx_audmix_be_hw_params,
+};
+
+static int imx_audmix_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *audmix_np = NULL, *out_cpu_np = NULL;
+ struct platform_device *audmix_pdev = NULL;
+ struct platform_device *cpu_pdev;
+ struct of_phandle_args args;
+ struct imx_audmix *priv;
+ int i, num_dai, ret;
+ const char *fe_name_pref = "HiFi-AUDMIX-FE-";
+ char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name;
+
+ if (pdev->dev.parent) {
+ audmix_np = pdev->dev.parent->of_node;
+ } else {
+ dev_err(&pdev->dev, "Missing parent device.\n");
+ return -EINVAL;
+ }
+
+ if (!audmix_np) {
+ dev_err(&pdev->dev, "Missign DT node for parent device.\n");
+ return -EINVAL;
+ }
+
+ audmix_pdev = of_find_device_by_node(audmix_np);
+ if (!audmix_pdev) {
+ dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n",
+ np->full_name);
+ return -EINVAL;
+ }
+
+ num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL);
+ if (num_dai != FSL_AUDMIX_MAX_DAIS) {
+ dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n",
+ audmix_np->full_name);
+ return -EINVAL;
+ }
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->num_dai = 2 * num_dai;
+ priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai *
+ sizeof(struct snd_soc_dai_link), GFP_KERNEL);
+ if (!priv->dai)
+ return -ENOMEM;
+
+ priv->num_dai_conf = num_dai;
+ priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf *
+ sizeof(struct snd_soc_codec_conf),
+ GFP_KERNEL);
+ if (!priv->dai_conf)
+ return -ENOMEM;
+
+ priv->num_dapm_routes = 3 * num_dai;
+ priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes *
+ sizeof(struct snd_soc_dapm_route),
+ GFP_KERNEL);
+ if (!priv->dapm_routes)
+ return -ENOMEM;
+
+ for (i = 0; i < num_dai; i++) {
+ ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i,
+ &args);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n");
+ return ret;
+ }
+
+ cpu_pdev = of_find_device_by_node(args.np);
+ if (!cpu_pdev) {
+ dev_err(&pdev->dev, "failed to find SAI platform device\n");
+ return -EINVAL;
+ }
+
+ dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s",
+ fe_name_pref, args.np->full_name + 1);
+
+ dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name);
+
+ if (i == 0) {
+ out_cpu_np = args.np;
+ capture_dai_name =
+ devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+ dai_name, "CPU-Capture");
+ }
+
+ priv->dai[i].name = dai_name;
+ priv->dai[i].stream_name = "HiFi-AUDMIX-FE";
+ priv->dai[i].codec_dai_name = "snd-soc-dummy-dai";
+ priv->dai[i].codec_name = "snd-soc-dummy";
+ priv->dai[i].cpu_of_node = args.np;
+ priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev);
+ priv->dai[i].platform_of_node = args.np;
+ priv->dai[i].dynamic = 1;
+ priv->dai[i].dpcm_playback = 1;
+ priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0);
+ priv->dai[i].ignore_pmdown_time = 1;
+ priv->dai[i].ops = &imx_audmix_fe_ops;
+
+ /* Add AUDMIX Backend */
+ be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+ "audmix-%d", i);
+ be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+ "AUDMIX-Playback-%d", i);
+ be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL,
+ "AUDMIX-Capture-%d", i);
+
+ priv->dai[num_dai + i].name = be_name;
+ priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai";
+ priv->dai[num_dai + i].codec_name = "snd-soc-dummy";
+ priv->dai[num_dai + i].cpu_of_node = audmix_np;
+ priv->dai[num_dai + i].cpu_dai_name = be_name;
+ priv->dai[num_dai + i].platform_name = "snd-soc-dummy";
+ priv->dai[num_dai + i].no_pcm = 1;
+ priv->dai[num_dai + i].dpcm_playback = 1;
+ priv->dai[num_dai + i].dpcm_capture = 1;
+ priv->dai[num_dai + i].ignore_pmdown_time = 1;
+ priv->dai[num_dai + i].ops = &imx_audmix_be_ops;
+
+ priv->dai_conf[i].of_node = args.np;
+ priv->dai_conf[i].name_prefix = dai_name;
+
+ priv->dapm_routes[i].source =
+ devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+ dai_name, "CPU-Playback");
+ priv->dapm_routes[i].sink = be_pb;
+ priv->dapm_routes[num_dai + i].source = be_pb;
+ priv->dapm_routes[num_dai + i].sink = be_cp;
+ priv->dapm_routes[2 * num_dai + i].source = be_cp;
+ priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name;
+ }
+
+ cpu_pdev = of_find_device_by_node(out_cpu_np);
+ if (!cpu_pdev) {
+ dev_err(&pdev->dev, "failed to find SAI platform device\n");
+ return -EINVAL;
+ }
+ priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1");
+ if (IS_ERR(priv->cpu_mclk)) {
+ ret = PTR_ERR(priv->cpu_mclk);
+ dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
+ return -EINVAL;
+ }
+
+ priv->audmix_pdev = audmix_pdev;
+ priv->out_pdev = cpu_pdev;
+
+ priv->card.dai_link = priv->dai;
+ priv->card.num_links = priv->num_dai;
+ priv->card.codec_conf = priv->dai_conf;
+ priv->card.num_configs = priv->num_dai_conf;
+ priv->card.dapm_routes = priv->dapm_routes;
+ priv->card.num_dapm_routes = priv->num_dapm_routes;
+ priv->card.dev = pdev->dev.parent;
+ priv->card.owner = THIS_MODULE;
+ priv->card.name = "imx-audmix";
+
+ platform_set_drvdata(pdev, &priv->card);
+ snd_soc_card_set_drvdata(&priv->card, priv);
+
+ ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct platform_driver imx_audmix_driver = {
+ .probe = imx_audmix_probe,
+ .driver = {
+ .name = "imx-audmix",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(imx_audmix_driver);
+
+MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver");
+MODULE_AUTHOR("Viorel Suman <[email protected]>");
+MODULE_ALIAS("platform:imx-audmix");
+MODULE_LICENSE("GPL v2");
--
2.7.4
This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
The Audio Mixer is a on-chip functional module that allows mixing of
two audio streams into a single audio stream.
Audio Mixer datasheet is available here:
https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf
Signed-off-by: Viorel Suman <[email protected]>
Acked-by: Nicolin Chen <[email protected]>
---
sound/soc/fsl/Kconfig | 7 +
sound/soc/fsl/Makefile | 3 +
sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++++++++++++++++++++++++++
sound/soc/fsl/fsl_audmix.h | 102 ++++++++
4 files changed, 690 insertions(+)
create mode 100644 sound/soc/fsl/fsl_audmix.c
create mode 100644 sound/soc/fsl/fsl_audmix.h
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7b1d997..0af2e056 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI
This option is only useful for out-of-tree drivers since
in-tree drivers select it automatically.
+config SND_SOC_FSL_AUDMIX
+ tristate "Audio Mixer (AUDMIX) module support"
+ select REGMAP_MMIO
+ help
+ Say Y if you want to add Audio Mixer (AUDMIX)
+ support for the NXP iMX CPUs.
+
config SND_SOC_FSL_SSI
tristate "Synchronous Serial Interface module (SSI) support"
select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 3c0ff31..4172d5a 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-audmix-objs := fsl_audmix.o
snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
snd-soc-fsl-sai-objs := fsl_sai.o
@@ -22,6 +23,8 @@ snd-soc-fsl-esai-objs := fsl_esai.o
snd-soc-fsl-micfil-objs := fsl_micfil.o
snd-soc-fsl-utils-objs := fsl_utils.o
snd-soc-fsl-dma-objs := fsl_dma.o
+
+obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o
obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
new file mode 100644
index 0000000..07b72a3
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -0,0 +1,578 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright 2017 NXP
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_audmix.h"
+
+#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \
+ SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts)
+
+static const char
+ *tdm_sel[] = { "TDM1", "TDM2", },
+ *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", },
+ *width_sel[] = { "16b", "18b", "20b", "24b", "32b", },
+ *endis_sel[] = { "Disabled", "Enabled", },
+ *updn_sel[] = { "Downward", "Upward", },
+ *mask_sel[] = { "Unmask", "Mask", };
+
+static const struct soc_enum fsl_audmix_enum[] = {
+/* FSL_AUDMIX_CTR enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MIXCLK_SHIFT, tdm_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTSRC_SHIFT, mode_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTWIDTH_SHIFT, width_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKRTDF_SHIFT, mask_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKCKDF_SHIFT, mask_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCMODE_SHIFT, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCSRC_SHIFT, tdm_sel),
+/* FSL_AUDMIX_ATCR0 enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 0, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 1, updn_sel),
+/* FSL_AUDMIX_ATCR1 enums */
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 0, endis_sel),
+SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 1, updn_sel),
+};
+
+struct fsl_audmix_state {
+ u8 tdms;
+ u8 clk;
+ char msg[64];
+};
+
+static const struct fsl_audmix_state prms[4][4] = {{
+ /* DIS->DIS, do nothing */
+ { .tdms = 0, .clk = 0, .msg = "" },
+ /* DIS->TDM1*/
+ { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" },
+ /* DIS->TDM2*/
+ { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" },
+ /* DIS->MIX */
+ { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" }
+}, { /* TDM1->DIS */
+ { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" },
+ /* TDM1->TDM1, do nothing */
+ { .tdms = 0, .clk = 0, .msg = "" },
+ /* TDM1->TDM2 */
+ { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" },
+ /* TDM1->MIX */
+ { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" }
+}, { /* TDM2->DIS */
+ { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" },
+ /* TDM2->TDM1 */
+ { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" },
+ /* TDM2->TDM2, do nothing */
+ { .tdms = 0, .clk = 0, .msg = "" },
+ /* TDM2->MIX */
+ { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" }
+}, { /* MIX->DIS */
+ { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" },
+ /* MIX->TDM1 */
+ { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" },
+ /* MIX->TDM2 */
+ { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" },
+ /* MIX->MIX, do nothing */
+ { .tdms = 0, .clk = 0, .msg = "" }
+}, };
+
+static int fsl_audmix_state_trans(struct snd_soc_component *comp,
+ unsigned int *mask, unsigned int *ctr,
+ const struct fsl_audmix_state prm)
+{
+ struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+ /* Enforce all required TDMs are started */
+ if ((priv->tdms & prm.tdms) != prm.tdms) {
+ dev_dbg(comp->dev, prm.msg);
+ return -EINVAL;
+ }
+
+ switch (prm.clk) {
+ case 1:
+ case 2:
+ /* Set mix clock */
+ (*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK;
+ (*ctr) |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int *item = ucontrol->value.enumerated.item;
+ unsigned int reg_val, val, mix_clk;
+ int ret = 0;
+
+ /* Get current state */
+ ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val);
+ if (ret)
+ return ret;
+
+ mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK)
+ >> FSL_AUDMIX_CTR_MIXCLK_SHIFT);
+ val = snd_soc_enum_item_to_val(e, item[0]);
+
+ dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val);
+
+ /**
+ * Ensure the current selected mixer clock is available
+ * for configuration propagation
+ */
+ if (!(priv->tdms & BIT(mix_clk))) {
+ dev_err(comp->dev,
+ "Started TDM%d needed for config propagation!\n",
+ mix_clk + 1);
+ return -EINVAL;
+ }
+
+ if (!(priv->tdms & BIT(val))) {
+ dev_err(comp->dev,
+ "The selected clock source has no TDM%d enabled!\n",
+ val + 1);
+ return -EINVAL;
+ }
+
+ return snd_soc_put_enum_double(kcontrol, ucontrol);
+}
+
+static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int *item = ucontrol->value.enumerated.item;
+ u32 out_src, mix_clk;
+ unsigned int reg_val, val, mask = 0, ctr = 0;
+ int ret = 0;
+
+ /* Get current state */
+ ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val);
+ if (ret)
+ return ret;
+
+ /* "From" state */
+ out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK)
+ >> FSL_AUDMIX_CTR_OUTSRC_SHIFT);
+ mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK)
+ >> FSL_AUDMIX_CTR_MIXCLK_SHIFT);
+
+ /* "To" state */
+ val = snd_soc_enum_item_to_val(e, item[0]);
+
+ dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val);
+
+ /* Check if state is changing ... */
+ if (out_src == val)
+ return 0;
+ /**
+ * Ensure the current selected mixer clock is available
+ * for configuration propagation
+ */
+ if (!(priv->tdms & BIT(mix_clk))) {
+ dev_err(comp->dev,
+ "Started TDM%d needed for config propagation!\n",
+ mix_clk + 1);
+ return -EINVAL;
+ }
+
+ /* Check state transition constraints */
+ ret = fsl_audmix_state_trans(comp, &mask, &ctr, prms[out_src][val]);
+ if (ret)
+ return ret;
+
+ /* Complete transition to new state */
+ mask |= FSL_AUDMIX_CTR_OUTSRC_MASK;
+ ctr |= FSL_AUDMIX_CTR_OUTSRC(val);
+
+ return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr);
+}
+
+static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = {
+ /* FSL_AUDMIX_CTR controls */
+ SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0],
+ snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src),
+ SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1],
+ snd_soc_get_enum_double, fsl_audmix_put_out_src),
+ SOC_ENUM("Output Width", fsl_audmix_enum[2]),
+ SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]),
+ SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]),
+ SOC_ENUM("Sync Mode Config", fsl_audmix_enum[5]),
+ SOC_ENUM("Sync Mode Clk Source", fsl_audmix_enum[6]),
+ /* TDM1 Attenuation controls */
+ SOC_ENUM("TDM1 Attenuation", fsl_audmix_enum[7]),
+ SOC_ENUM("TDM1 Attenuation Direction", fsl_audmix_enum[8]),
+ SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AUDMIX_ATCR0,
+ 2, 0x00fff, 0),
+ SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AUDMIX_ATIVAL0,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP0,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN0,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT0,
+ 0, 0x3ffff, 0),
+ /* TDM2 Attenuation controls */
+ SOC_ENUM("TDM2 Attenuation", fsl_audmix_enum[9]),
+ SOC_ENUM("TDM2 Attenuation Direction", fsl_audmix_enum[10]),
+ SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AUDMIX_ATCR1,
+ 2, 0x00fff, 0),
+ SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AUDMIX_ATIVAL1,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP1,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN1,
+ 0, 0x3ffff, 0),
+ SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT1,
+ 0, 0x3ffff, 0),
+};
+
+static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *comp = dai->component;
+ u32 mask = 0, ctr = 0;
+
+ /* AUDMIX is working in DSP_A format only */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* For playback the AUDMIX is slave, and for record is master */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ /* Output data will be written on positive edge of the clock */
+ ctr |= FSL_AUDMIX_CTR_OUTCKPOL(0);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ /* Output data will be written on negative edge of the clock */
+ ctr |= FSL_AUDMIX_CTR_OUTCKPOL(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mask |= FSL_AUDMIX_CTR_OUTCKPOL_MASK;
+
+ return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr);
+}
+
+static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai);
+
+ /* Capture stream shall not be handled */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ priv->tdms |= BIT(dai->driver->id);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ priv->tdms &= ~BIT(dai->driver->id);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_audmix_dai_ops = {
+ .set_fmt = fsl_audmix_dai_set_fmt,
+ .trigger = fsl_audmix_dai_trigger,
+};
+
+static struct snd_soc_dai_driver fsl_audmix_dai[] = {
+ {
+ .id = 0,
+ .name = "audmix-0",
+ .playback = {
+ .stream_name = "AUDMIX-Playback-0",
+ .channels_min = 8,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_AUDMIX_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AUDMIX-Capture-0",
+ .channels_min = 8,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_AUDMIX_FORMATS,
+ },
+ .ops = &fsl_audmix_dai_ops,
+ },
+ {
+ .id = 1,
+ .name = "audmix-1",
+ .playback = {
+ .stream_name = "AUDMIX-Playback-1",
+ .channels_min = 8,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_AUDMIX_FORMATS,
+ },
+ .capture = {
+ .stream_name = "AUDMIX-Capture-1",
+ .channels_min = 8,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_AUDMIX_FORMATS,
+ },
+ .ops = &fsl_audmix_dai_ops,
+ },
+};
+
+static const struct snd_soc_component_driver fsl_audmix_component = {
+ .name = "fsl-audmix-dai",
+ .controls = fsl_audmix_snd_controls,
+ .num_controls = ARRAY_SIZE(fsl_audmix_snd_controls),
+};
+
+static bool fsl_audmix_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_AUDMIX_CTR:
+ case FSL_AUDMIX_STR:
+ case FSL_AUDMIX_ATCR0:
+ case FSL_AUDMIX_ATIVAL0:
+ case FSL_AUDMIX_ATSTPUP0:
+ case FSL_AUDMIX_ATSTPDN0:
+ case FSL_AUDMIX_ATSTPTGT0:
+ case FSL_AUDMIX_ATTNVAL0:
+ case FSL_AUDMIX_ATSTP0:
+ case FSL_AUDMIX_ATCR1:
+ case FSL_AUDMIX_ATIVAL1:
+ case FSL_AUDMIX_ATSTPUP1:
+ case FSL_AUDMIX_ATSTPDN1:
+ case FSL_AUDMIX_ATSTPTGT1:
+ case FSL_AUDMIX_ATTNVAL1:
+ case FSL_AUDMIX_ATSTP1:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_audmix_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_AUDMIX_CTR:
+ case FSL_AUDMIX_ATCR0:
+ case FSL_AUDMIX_ATIVAL0:
+ case FSL_AUDMIX_ATSTPUP0:
+ case FSL_AUDMIX_ATSTPDN0:
+ case FSL_AUDMIX_ATSTPTGT0:
+ case FSL_AUDMIX_ATCR1:
+ case FSL_AUDMIX_ATIVAL1:
+ case FSL_AUDMIX_ATSTPUP1:
+ case FSL_AUDMIX_ATSTPDN1:
+ case FSL_AUDMIX_ATSTPTGT1:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct reg_default fsl_audmix_reg[] = {
+ { FSL_AUDMIX_CTR, 0x00060 },
+ { FSL_AUDMIX_STR, 0x00003 },
+ { FSL_AUDMIX_ATCR0, 0x00000 },
+ { FSL_AUDMIX_ATIVAL0, 0x3FFFF },
+ { FSL_AUDMIX_ATSTPUP0, 0x2AAAA },
+ { FSL_AUDMIX_ATSTPDN0, 0x30000 },
+ { FSL_AUDMIX_ATSTPTGT0, 0x00010 },
+ { FSL_AUDMIX_ATTNVAL0, 0x00000 },
+ { FSL_AUDMIX_ATSTP0, 0x00000 },
+ { FSL_AUDMIX_ATCR1, 0x00000 },
+ { FSL_AUDMIX_ATIVAL1, 0x3FFFF },
+ { FSL_AUDMIX_ATSTPUP1, 0x2AAAA },
+ { FSL_AUDMIX_ATSTPDN1, 0x30000 },
+ { FSL_AUDMIX_ATSTPTGT1, 0x00010 },
+ { FSL_AUDMIX_ATTNVAL1, 0x00000 },
+ { FSL_AUDMIX_ATSTP1, 0x00000 },
+};
+
+static const struct regmap_config fsl_audmix_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = FSL_AUDMIX_ATSTP1,
+ .reg_defaults = fsl_audmix_reg,
+ .num_reg_defaults = ARRAY_SIZE(fsl_audmix_reg),
+ .readable_reg = fsl_audmix_readable_reg,
+ .writeable_reg = fsl_audmix_writeable_reg,
+ .cache_type = REGCACHE_FLAT,
+};
+
+static const struct of_device_id fsl_audmix_ids[] = {
+ {
+ .compatible = "fsl,imx8qm-audmix",
+ .data = "imx-audmix",
+ },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_audmix_ids);
+
+static int fsl_audmix_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct fsl_audmix *priv;
+ struct resource *res;
+ const char *mdrv;
+ const struct of_device_id *of_id;
+ void __iomem *regs;
+ int ret;
+
+ of_id = of_match_device(fsl_audmix_ids, dev);
+ if (!of_id || !of_id->data)
+ return -EINVAL;
+
+ mdrv = of_id->data;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ /* Get the addresses */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ priv->regmap = devm_regmap_init_mmio_clk(dev, "ipg", regs,
+ &fsl_audmix_regmap_config);
+ if (IS_ERR(priv->regmap)) {
+ dev_err(dev, "failed to init regmap\n");
+ return PTR_ERR(priv->regmap);
+ }
+
+ priv->ipg_clk = devm_clk_get(dev, "ipg");
+ if (IS_ERR(priv->ipg_clk)) {
+ dev_err(dev, "failed to get ipg clock\n");
+ return PTR_ERR(priv->ipg_clk);
+ }
+
+ platform_set_drvdata(pdev, priv);
+ pm_runtime_enable(dev);
+
+ ret = devm_snd_soc_register_component(dev, &fsl_audmix_component,
+ fsl_audmix_dai,
+ ARRAY_SIZE(fsl_audmix_dai));
+ if (ret) {
+ dev_err(dev, "failed to register ASoC DAI\n");
+ return ret;
+ }
+
+ priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0);
+ if (IS_ERR(priv->pdev)) {
+ ret = PTR_ERR(priv->pdev);
+ dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret);
+ }
+
+ return ret;
+}
+
+static int fsl_audmix_remove(struct platform_device *pdev)
+{
+ struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev);
+
+ if (priv->pdev)
+ platform_device_unregister(priv->pdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int fsl_audmix_runtime_resume(struct device *dev)
+{
+ struct fsl_audmix *priv = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(priv->ipg_clk);
+ if (ret) {
+ dev_err(dev, "Failed to enable IPG clock: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(priv->regmap, false);
+ regcache_mark_dirty(priv->regmap);
+
+ return regcache_sync(priv->regmap);
+}
+
+static int fsl_audmix_runtime_suspend(struct device *dev)
+{
+ struct fsl_audmix *priv = dev_get_drvdata(dev);
+
+ regcache_cache_only(priv->regmap, true);
+
+ clk_disable_unprepare(priv->ipg_clk);
+
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static const struct dev_pm_ops fsl_audmix_pm = {
+ SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend,
+ fsl_audmix_runtime_resume,
+ NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+ pm_runtime_force_resume)
+};
+
+static struct platform_driver fsl_audmix_driver = {
+ .probe = fsl_audmix_probe,
+ .remove = fsl_audmix_remove,
+ .driver = {
+ .name = "fsl-audmix",
+ .of_match_table = fsl_audmix_ids,
+ .pm = &fsl_audmix_pm,
+ },
+};
+module_platform_driver(fsl_audmix_driver);
+
+MODULE_DESCRIPTION("NXP AUDMIX ASoC DAI driver");
+MODULE_AUTHOR("Viorel Suman <[email protected]>");
+MODULE_ALIAS("platform:fsl-audmix");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h
new file mode 100644
index 0000000..7812ffe
--- /dev/null
+++ b/sound/soc/fsl/fsl_audmix.h
@@ -0,0 +1,102 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright 2017 NXP
+ */
+
+#ifndef __FSL_AUDMIX_H
+#define __FSL_AUDMIX_H
+
+#define FSL_AUDMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+/* AUDMIX Registers */
+#define FSL_AUDMIX_CTR 0x200 /* Control */
+#define FSL_AUDMIX_STR 0x204 /* Status */
+
+#define FSL_AUDMIX_ATCR0 0x208 /* Attenuation Control */
+#define FSL_AUDMIX_ATIVAL0 0x20c /* Attenuation Initial Value */
+#define FSL_AUDMIX_ATSTPUP0 0x210 /* Attenuation step up factor */
+#define FSL_AUDMIX_ATSTPDN0 0x214 /* Attenuation step down factor */
+#define FSL_AUDMIX_ATSTPTGT0 0x218 /* Attenuation step target */
+#define FSL_AUDMIX_ATTNVAL0 0x21c /* Attenuation Value */
+#define FSL_AUDMIX_ATSTP0 0x220 /* Attenuation step number */
+
+#define FSL_AUDMIX_ATCR1 0x228 /* Attenuation Control */
+#define FSL_AUDMIX_ATIVAL1 0x22c /* Attenuation Initial Value */
+#define FSL_AUDMIX_ATSTPUP1 0x230 /* Attenuation step up factor */
+#define FSL_AUDMIX_ATSTPDN1 0x234 /* Attenuation step down factor */
+#define FSL_AUDMIX_ATSTPTGT1 0x238 /* Attenuation step target */
+#define FSL_AUDMIX_ATTNVAL1 0x23c /* Attenuation Value */
+#define FSL_AUDMIX_ATSTP1 0x240 /* Attenuation step number */
+
+/* AUDMIX Control Register */
+#define FSL_AUDMIX_CTR_MIXCLK_SHIFT 0
+#define FSL_AUDMIX_CTR_MIXCLK_MASK BIT(FSL_AUDMIX_CTR_MIXCLK_SHIFT)
+#define FSL_AUDMIX_CTR_MIXCLK(i) ((i) << FSL_AUDMIX_CTR_MIXCLK_SHIFT)
+#define FSL_AUDMIX_CTR_OUTSRC_SHIFT 1
+#define FSL_AUDMIX_CTR_OUTSRC_MASK (0x3 << FSL_AUDMIX_CTR_OUTSRC_SHIFT)
+#define FSL_AUDMIX_CTR_OUTSRC(i) (((i) << FSL_AUDMIX_CTR_OUTSRC_SHIFT)\
+ & FSL_AUDMIX_CTR_OUTSRC_MASK)
+#define FSL_AUDMIX_CTR_OUTWIDTH_SHIFT 3
+#define FSL_AUDMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)
+#define FSL_AUDMIX_CTR_OUTWIDTH(i) (((i) << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)\
+ & FSL_AUDMIX_CTR_OUTWIDTH_MASK)
+#define FSL_AUDMIX_CTR_OUTCKPOL_SHIFT 6
+#define FSL_AUDMIX_CTR_OUTCKPOL_MASK BIT(FSL_AUDMIX_CTR_OUTCKPOL_SHIFT)
+#define FSL_AUDMIX_CTR_OUTCKPOL(i) ((i) << FSL_AUDMIX_CTR_OUTCKPOL_SHIFT)
+#define FSL_AUDMIX_CTR_MASKRTDF_SHIFT 7
+#define FSL_AUDMIX_CTR_MASKRTDF_MASK BIT(FSL_AUDMIX_CTR_MASKRTDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKRTDF(i) ((i) << FSL_AUDMIX_CTR_MASKRTDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKCKDF_SHIFT 8
+#define FSL_AUDMIX_CTR_MASKCKDF_MASK BIT(FSL_AUDMIX_CTR_MASKCKDF_SHIFT)
+#define FSL_AUDMIX_CTR_MASKCKDF(i) ((i) << FSL_AUDMIX_CTR_MASKCKDF_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCMODE_SHIFT 9
+#define FSL_AUDMIX_CTR_SYNCMODE_MASK BIT(FSL_AUDMIX_CTR_SYNCMODE_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCMODE(i) ((i) << FSL_AUDMIX_CTR_SYNCMODE_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCSRC_SHIFT 10
+#define FSL_AUDMIX_CTR_SYNCSRC_MASK BIT(FSL_AUDMIX_CTR_SYNCSRC_SHIFT)
+#define FSL_AUDMIX_CTR_SYNCSRC(i) ((i) << FSL_AUDMIX_CTR_SYNCSRC_SHIFT)
+
+/* AUDMIX Status Register */
+#define FSL_AUDMIX_STR_RATEDIFF BIT(0)
+#define FSL_AUDMIX_STR_CLKDIFF BIT(1)
+#define FSL_AUDMIX_STR_MIXSTAT_SHIFT 2
+#define FSL_AUDMIX_STR_MIXSTAT_MASK (0x3 << FSL_AUDMIX_STR_MIXSTAT_SHIFT)
+#define FSL_AUDMIX_STR_MIXSTAT(i) (((i) & FSL_AUDMIX_STR_MIXSTAT_MASK) \
+ >> FSL_AUDMIX_STR_MIXSTAT_SHIFT)
+/* AUDMIX Attenuation Control Register */
+#define FSL_AUDMIX_ATCR_AT_EN BIT(0)
+#define FSL_AUDMIX_ATCR_AT_UPDN BIT(1)
+#define FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT 2
+#define FSL_AUDMIX_ATCR_ATSTPDFI_MASK \
+ (0xfff << FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT)
+
+/* AUDMIX Attenuation Initial Value Register */
+#define FSL_AUDMIX_ATIVAL_ATINVAL_MASK 0x3FFFF
+
+/* AUDMIX Attenuation Step Up Factor Register */
+#define FSL_AUDMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF
+
+/* AUDMIX Attenuation Step Down Factor Register */
+#define FSL_AUDMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF
+
+/* AUDMIX Attenuation Step Target Register */
+#define FSL_AUDMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF
+
+/* AUDMIX Attenuation Value Register */
+#define FSL_AUDMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF
+
+/* AUDMIX Attenuation Step Number Register */
+#define FSL_AUDMIX_ATSTP_STPCTR_MASK 0x3FFFF
+
+#define FSL_AUDMIX_MAX_DAIS 2
+struct fsl_audmix {
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ struct clk *ipg_clk;
+ u8 tdms;
+};
+
+#endif /* __FSL_AUDMIX_H */
--
2.7.4
Hi Viorel,
Thank you for the patch! Perhaps something to improve:
[auto build test WARNING on asoc/for-next]
[also build test WARNING on v5.0 next-20190304]
[if your patch is applied to the wrong git tree, please drop us a note to help improve the system]
url: https://github.com/0day-ci/linux/commits/Viorel-Suman/Add-NXP-AUDMIX-device-and-machine-drivers/20190305-023038
base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
config: nds32-allyesconfig (attached as .config)
compiler: nds32le-linux-gcc (GCC) 6.4.0
reproduce:
wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
chmod +x ~/bin/make.cross
# save the attached .config to linux build tree
GCC_VERSION=6.4.0 make.cross ARCH=nds32
All warnings (new ones prefixed by >>):
In file included from include/linux/printk.h:331:0,
from include/linux/kernel.h:14,
from include/linux/clk.h:16,
from sound/soc/fsl/fsl_audmix.c:8:
sound/soc/fsl/fsl_audmix.c: In function 'fsl_audmix_state_trans':
include/linux/dynamic_debug.h:80:13: error: initializer element is not constant
.format = (fmt), \
^
include/linux/dynamic_debug.h:111:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA_KEY'
DEFINE_DYNAMIC_DEBUG_METADATA_KEY(name, fmt, 0, 0)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
include/linux/dynamic_debug.h:133:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA'
DEFINE_DYNAMIC_DEBUG_METADATA(descriptor, fmt); \
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
include/linux/device.h:1473:2: note: in expansion of macro 'dynamic_dev_dbg'
dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~
>> sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg'
dev_dbg(comp->dev, prm.msg);
^~~~~~~
include/linux/dynamic_debug.h:80:13: note: (near initialization for 'descriptor.format')
.format = (fmt), \
^
include/linux/dynamic_debug.h:111:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA_KEY'
DEFINE_DYNAMIC_DEBUG_METADATA_KEY(name, fmt, 0, 0)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
include/linux/dynamic_debug.h:133:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA'
DEFINE_DYNAMIC_DEBUG_METADATA(descriptor, fmt); \
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
include/linux/device.h:1473:2: note: in expansion of macro 'dynamic_dev_dbg'
dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~
>> sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg'
dev_dbg(comp->dev, prm.msg);
^~~~~~~
vim +/dev_dbg +93 sound/soc/fsl/fsl_audmix.c
85
86 static int fsl_audmix_state_trans(struct snd_soc_component *comp,
87 unsigned int *mask, unsigned int *ctr,
88 const struct fsl_audmix_state prm)
89 {
90 struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp);
91 /* Enforce all required TDMs are started */
92 if ((priv->tdms & prm.tdms) != prm.tdms) {
> 93 dev_dbg(comp->dev, prm.msg);
94 return -EINVAL;
95 }
96
97 switch (prm.clk) {
98 case 1:
99 case 2:
100 /* Set mix clock */
101 (*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK;
102 (*ctr) |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1);
103 break;
104 default:
105 break;
106 }
107
108 return 0;
109 }
110
---
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https://lists.01.org/pipermail/kbuild-all Intel Corporation