1 files changed, 272 insertions, 0 deletions
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
new file mode 100644
index 000000000..1bbd1d077
--- /dev/null
+++ b/sound/soc/soc-utils.c
@@ -0,0 +1,272 @@
+// SPDX-License-Identifier: GPL-2.0+
+//
+// soc-util.c  --  ALSA SoC Audio Layer utility functions
+//
+// Copyright 2009 Wolfson Microelectronics PLC.
+//
+// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+//         Liam Girdwood <lrg@slimlogic.co.uk>
+
+#include <linux/platform_device.h>
+#include <linux/export.h>
+#include <linux/math.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots)
+{
+	return sample_size * channels * tdm_slots;
+}
+EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size);
+
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
+{
+	int sample_size;
+
+	sample_size = snd_pcm_format_width(params_format(params));
+	if (sample_size < 0)
+		return sample_size;
+
+	return snd_soc_calc_frame_size(sample_size, params_channels(params),
+				       1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size);
+
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
+{
+	return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots);
+}
+EXPORT_SYMBOL_GPL(snd_soc_calc_bclk);
+
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
+{
+	int ret;
+
+	ret = snd_soc_params_to_frame_size(params);
+
+	if (ret > 0)
+		return ret * params_rate(params);
+	else
+		return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
+
+/**
+ * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info.
+ *
+ * Calculate the bclk from the params sample rate, the tdm slot count and the
+ * tdm slot width. Optionally round-up the slot count to a given multiple.
+ * Either or both of tdm_width and tdm_slots can be 0.
+ *
+ * If tdm_width == 0:	use params_width() as the slot width.
+ * If tdm_slots == 0:	use params_channels() as the slot count.
+ *
+ * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0)
+ * will be rounded up to a multiple of slot_multiple. This is mainly useful for
+ * I2S mode, which has a left and right phase so the number of slots is always
+ * a multiple of 2.
+ *
+ * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent
+ * to calling snd_soc_params_to_bclk().
+ *
+ * @params:        Pointer to struct_pcm_hw_params.
+ * @tdm_width:     Width in bits of the tdm slots. Must be >= 0.
+ * @tdm_slots:     Number of tdm slots per frame. Must be >= 0.
+ * @slot_multiple: If >1 roundup slot count to a multiple of this value.
+ *
+ * Return: bclk frequency in Hz, else a negative error code if params format
+ *	   is invalid.
+ */
+int snd_soc_tdm_params_to_bclk(struct snd_pcm_hw_params *params,
+			       int tdm_width, int tdm_slots, int slot_multiple)
+{
+	if (!tdm_slots)
+		tdm_slots = params_channels(params);
+
+	if (slot_multiple > 1)
+		tdm_slots = roundup(tdm_slots, slot_multiple);
+
+	if (!tdm_width) {
+		tdm_width = snd_pcm_format_width(params_format(params));
+		if (tdm_width < 0)
+			return tdm_width;
+	}
+
+	return snd_soc_calc_bclk(params_rate(params), tdm_width, 1, tdm_slots);
+}
+EXPORT_SYMBOL_GPL(snd_soc_tdm_params_to_bclk);
+
+static const struct snd_pcm_hardware dummy_dma_hardware = {
+	/* Random values to keep userspace happy when checking constraints */
+	.info			= SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_BLOCK_TRANSFER,
+	.buffer_bytes_max	= 128*1024,
+	.period_bytes_min	= PAGE_SIZE,
+	.period_bytes_max	= PAGE_SIZE*2,
+	.periods_min		= 2,
+	.periods_max		= 128,
+};
+
+
+static const struct snd_soc_component_driver dummy_platform;
+
+static int dummy_dma_open(struct snd_soc_component *component,
+			  struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	int i;
+
+	/*
+	 * If there are other components associated with rtd, we shouldn't
+	 * override their hwparams
+	 */
+	for_each_rtd_components(rtd, i, component) {
+		if (component->driver == &dummy_platform)
+			return 0;
+	}
+
+	/* BE's dont need dummy params */
+	if (!rtd->dai_link->no_pcm)
+		snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver dummy_platform = {
+	.open		= dummy_dma_open,
+};
+
+static const struct snd_soc_component_driver dummy_codec = {
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+};
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_384000
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_S24_3LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+/*
+ * Select these from Sound Card Manually
+ *	SND_SOC_POSSIBLE_DAIFMT_CBP_CFP
+ *	SND_SOC_POSSIBLE_DAIFMT_CBP_CFC
+ *	SND_SOC_POSSIBLE_DAIFMT_CBC_CFP
+ *	SND_SOC_POSSIBLE_DAIFMT_CBC_CFC
+ */
+static u64 dummy_dai_formats =
+	SND_SOC_POSSIBLE_DAIFMT_I2S	|
+	SND_SOC_POSSIBLE_DAIFMT_RIGHT_J	|
+	SND_SOC_POSSIBLE_DAIFMT_LEFT_J	|
+	SND_SOC_POSSIBLE_DAIFMT_DSP_A	|
+	SND_SOC_POSSIBLE_DAIFMT_DSP_B	|
+	SND_SOC_POSSIBLE_DAIFMT_AC97	|
+	SND_SOC_POSSIBLE_DAIFMT_PDM	|
+	SND_SOC_POSSIBLE_DAIFMT_GATED	|
+	SND_SOC_POSSIBLE_DAIFMT_CONT	|
+	SND_SOC_POSSIBLE_DAIFMT_NB_NF	|
+	SND_SOC_POSSIBLE_DAIFMT_NB_IF	|
+	SND_SOC_POSSIBLE_DAIFMT_IB_NF	|
+	SND_SOC_POSSIBLE_DAIFMT_IB_IF;
+
+static const struct snd_soc_dai_ops dummy_dai_ops = {
+	.auto_selectable_formats	= &dummy_dai_formats,
+	.num_auto_selectable_formats	= 1,
+};
+
+/*
+ * The dummy CODEC is only meant to be used in situations where there is no
+ * actual hardware.
+ *
+ * If there is actual hardware even if it does not have a control bus
+ * the hardware will still have constraints like supported samplerates, etc.
+ * which should be modelled. And the data flow graph also should be modelled
+ * using DAPM.
+ */
+static struct snd_soc_dai_driver dummy_dai = {
+	.name = "snd-soc-dummy-dai",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates = STUB_RATES,
+		.formats = STUB_FORMATS,
+	 },
+	.ops = &dummy_dai_ops,
+};
+
+int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
+{
+	if (dai->driver == &dummy_dai)
+		return 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy);
+
+int snd_soc_component_is_dummy(struct snd_soc_component *component)
+{
+	return ((component->driver == &dummy_platform) ||
+		(component->driver == &dummy_codec));
+}
+
+static int snd_soc_dummy_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &dummy_codec, &dummy_dai, 1);
+	if (ret < 0)
+		return ret;
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform,
+					      NULL, 0);
+
+	return ret;
+}
+
+static struct platform_driver soc_dummy_driver = {
+	.driver = {
+		.name = "snd-soc-dummy",
+	},
+	.probe = snd_soc_dummy_probe,
+};
+
+static struct platform_device *soc_dummy_dev;
+
+int __init snd_soc_util_init(void)
+{
+	int ret;
+
+	soc_dummy_dev =
+		platform_device_register_simple("snd-soc-dummy", -1, NULL, 0);
+	if (IS_ERR(soc_dummy_dev))
+		return PTR_ERR(soc_dummy_dev);
+
+	ret = platform_driver_register(&soc_dummy_driver);
+	if (ret != 0)
+		platform_device_unregister(soc_dummy_dev);
+
+	return ret;
+}
+
+void snd_soc_util_exit(void)
+{
+	platform_driver_unregister(&soc_dummy_driver);
+	platform_device_unregister(soc_dummy_dev);
+}