Adding upstream version 4.19.249.upstream/4.19.249upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 388 insertions, 0 deletions

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 000000000..f5d700411
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,388 @@
+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * linux/sound/soc-dai.h -- ALSA SoC Layer
+ *
+ * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
+ *
+ * Digital Audio Interface (DAI) API.
+ */
+
+#ifndef __LINUX_SND_SOC_DAI_H
+#define __LINUX_SND_SOC_DAI_H
+
+
+#include <linux/list.h>
+#include <sound/asoc.h>
+
+struct snd_pcm_substream;
+struct snd_soc_dapm_widget;
+struct snd_compr_stream;
+
+/*
+ * DAI hardware audio formats.
+ *
+ * Describes the physical PCM data formating and clocking. Add new formats
+ * to the end.
+ */
+#define SND_SOC_DAIFMT_I2S		SND_SOC_DAI_FORMAT_I2S
+#define SND_SOC_DAIFMT_RIGHT_J		SND_SOC_DAI_FORMAT_RIGHT_J
+#define SND_SOC_DAIFMT_LEFT_J		SND_SOC_DAI_FORMAT_LEFT_J
+#define SND_SOC_DAIFMT_DSP_A		SND_SOC_DAI_FORMAT_DSP_A
+#define SND_SOC_DAIFMT_DSP_B		SND_SOC_DAI_FORMAT_DSP_B
+#define SND_SOC_DAIFMT_AC97		SND_SOC_DAI_FORMAT_AC97
+#define SND_SOC_DAIFMT_PDM		SND_SOC_DAI_FORMAT_PDM
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
+#define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J
+
+/*
+ * DAI Clock gating.
+ *
+ * DAI bit clocks can be be gated (disabled) when the DAI is not
+ * sending or receiving PCM data in a frame. This can be used to save power.
+ */
+#define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
+
+/*
+ * DAI hardware signal polarity.
+ *
+ * Specifies whether the DAI can also support inverted clocks for the specified
+ * format.
+ *
+ * BCLK:
+ * - "normal" polarity means signal is available at rising edge of BCLK
+ * - "inverted" polarity means signal is available at falling edge of BCLK
+ *
+ * FSYNC "normal" polarity depends on the frame format:
+ * - I2S: frame consists of left then right channel data. Left channel starts
+ *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
+ * - Left/Right Justified: frame consists of left then right channel data.
+ *      Left channel starts with rising FSYNC edge, right channel starts with
+ *      falling FSYNC edge.
+ * - DSP A/B: Frame starts with rising FSYNC edge.
+ * - AC97: Frame starts with rising FSYNC edge.
+ *
+ * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
+ */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */
+
+/*
+ * DAI hardware clock masters.
+ *
+ * This is wrt the codec, the inverse is true for the interface
+ * i.e. if the codec is clk and FRM master then the interface is
+ * clk and frame slave.
+ */
+#define SND_SOC_DAIFMT_CBM_CFM		(1 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk slave & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM slave */
+
+#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
+#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
+#define SND_SOC_DAIFMT_INV_MASK		0x0f00
+#define SND_SOC_DAIFMT_MASTER_MASK	0xf000
+
+/*
+ * Master Clock Directions
+ */
+#define SND_SOC_CLOCK_IN		0
+#define SND_SOC_CLOCK_OUT		1
+
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+			       SNDRV_PCM_FMTBIT_S16_LE |\
+			       SNDRV_PCM_FMTBIT_S16_BE |\
+			       SNDRV_PCM_FMTBIT_S20_3LE |\
+			       SNDRV_PCM_FMTBIT_S20_3BE |\
+			       SNDRV_PCM_FMTBIT_S20_LE |\
+			       SNDRV_PCM_FMTBIT_S20_BE |\
+			       SNDRV_PCM_FMTBIT_S24_3LE |\
+			       SNDRV_PCM_FMTBIT_S24_3BE |\
+                               SNDRV_PCM_FMTBIT_S32_LE |\
+                               SNDRV_PCM_FMTBIT_S32_BE)
+
+struct snd_soc_dai_driver;
+struct snd_soc_dai;
+struct snd_ac97_bus_ops;
+
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+	unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+	int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
+
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
+
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+	unsigned int tx_num, unsigned int *tx_slot,
+	unsigned int rx_num, unsigned int *rx_slot);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction);
+
+
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+		unsigned int *tx_num, unsigned int *tx_slot,
+		unsigned int *rx_num, unsigned int *rx_slot);
+
+int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
+
+struct snd_soc_dai_ops {
+	/*
+	 * DAI clocking configuration, all optional.
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_sysclk)(struct snd_soc_dai *dai,
+		int clk_id, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+		unsigned int freq_in, unsigned int freq_out);
+	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+	int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
+
+	/*
+	 * DAI format configuration
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+	int (*xlate_tdm_slot_mask)(unsigned int slots,
+		unsigned int *tx_mask, unsigned int *rx_mask);
+	int (*set_tdm_slot)(struct snd_soc_dai *dai,
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width);
+	int (*set_channel_map)(struct snd_soc_dai *dai,
+		unsigned int tx_num, unsigned int *tx_slot,
+		unsigned int rx_num, unsigned int *rx_slot);
+	int (*get_channel_map)(struct snd_soc_dai *dai,
+			unsigned int *tx_num, unsigned int *tx_slot,
+			unsigned int *rx_num, unsigned int *rx_slot);
+	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+
+	int (*set_sdw_stream)(struct snd_soc_dai *dai,
+			void *stream, int direction);
+	/*
+	 * DAI digital mute - optional.
+	 * Called by soc-core to minimise any pops.
+	 */
+	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
+
+	/*
+	 * ALSA PCM audio operations - all optional.
+	 * Called by soc-core during audio PCM operations.
+	 */
+	int (*startup)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	void (*shutdown)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*hw_params)(struct snd_pcm_substream *,
+		struct snd_pcm_hw_params *, struct snd_soc_dai *);
+	int (*hw_free)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*prepare)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	/*
+	 * NOTE: Commands passed to the trigger function are not necessarily
+	 * compatible with the current state of the dai. For example this
+	 * sequence of commands is possible: START STOP STOP.
+	 * So do not unconditionally use refcounting functions in the trigger
+	 * function, e.g. clk_enable/disable.
+	 */
+	int (*trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
+	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
+	/*
+	 * For hardware based FIFO caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+};
+
+struct snd_soc_cdai_ops {
+	/*
+	 * for compress ops
+	 */
+	int (*startup)(struct snd_compr_stream *,
+			struct snd_soc_dai *);
+	int (*shutdown)(struct snd_compr_stream *,
+			struct snd_soc_dai *);
+	int (*set_params)(struct snd_compr_stream *,
+			struct snd_compr_params *, struct snd_soc_dai *);
+	int (*get_params)(struct snd_compr_stream *,
+			struct snd_codec *, struct snd_soc_dai *);
+	int (*set_metadata)(struct snd_compr_stream *,
+			struct snd_compr_metadata *, struct snd_soc_dai *);
+	int (*get_metadata)(struct snd_compr_stream *,
+			struct snd_compr_metadata *, struct snd_soc_dai *);
+	int (*trigger)(struct snd_compr_stream *, int,
+			struct snd_soc_dai *);
+	int (*pointer)(struct snd_compr_stream *,
+			struct snd_compr_tstamp *, struct snd_soc_dai *);
+	int (*ack)(struct snd_compr_stream *, size_t,
+			struct snd_soc_dai *);
+};
+
+/*
+ * Digital Audio Interface Driver.
+ *
+ * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
+ * operations and capabilities. Codec and platform drivers will register this
+ * structure for every DAI they have.
+ *
+ * This structure covers the clocking, formating and ALSA operations for each
+ * interface.
+ */
+struct snd_soc_dai_driver {
+	/* DAI description */
+	const char *name;
+	unsigned int id;
+	unsigned int base;
+	struct snd_soc_dobj dobj;
+
+	/* DAI driver callbacks */
+	int (*probe)(struct snd_soc_dai *dai);
+	int (*remove)(struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
+	/* compress dai */
+	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
+	/* Optional Callback used at pcm creation*/
+	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
+		       struct snd_soc_dai *dai);
+	/* DAI is also used for the control bus */
+	bool bus_control;
+
+	/* ops */
+	const struct snd_soc_dai_ops *ops;
+	const struct snd_soc_cdai_ops *cops;
+
+	/* DAI capabilities */
+	struct snd_soc_pcm_stream capture;
+	struct snd_soc_pcm_stream playback;
+	unsigned int symmetric_rates:1;
+	unsigned int symmetric_channels:1;
+	unsigned int symmetric_samplebits:1;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+	const char *name;
+	int id;
+	struct device *dev;
+
+	/* driver ops */
+	struct snd_soc_dai_driver *driver;
+
+	/* DAI runtime info */
+	unsigned int capture_active;		/* stream usage count */
+	unsigned int playback_active;		/* stream usage count */
+	unsigned int probed:1;
+
+	unsigned int active;
+
+	struct snd_soc_dapm_widget *playback_widget;
+	struct snd_soc_dapm_widget *capture_widget;
+
+	/* DAI DMA data */
+	void *playback_dma_data;
+	void *capture_dma_data;
+
+	/* Symmetry data - only valid if symmetry is being enforced */
+	unsigned int rate;
+	unsigned int channels;
+	unsigned int sample_bits;
+
+	/* parent platform/codec */
+	struct snd_soc_component *component;
+
+	/* CODEC TDM slot masks and params (for fixup) */
+	unsigned int tx_mask;
+	unsigned int rx_mask;
+
+	struct list_head list;
+};
+
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+					     const struct snd_pcm_substream *ss)
+{
+	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		dai->playback_dma_data : dai->capture_dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+					    const struct snd_pcm_substream *ss,
+					    void *data)
+{
+	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->playback_dma_data = data;
+	else
+		dai->capture_dma_data = data;
+}
+
+static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
+					     void *playback, void *capture)
+{
+	dai->playback_dma_data = playback;
+	dai->capture_dma_data = capture;
+}
+
+static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
+		void *data)
+{
+	dev_set_drvdata(dai->dev, data);
+}
+
+static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
+{
+	return dev_get_drvdata(dai->dev);
+}
+
+/**
+ * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
+ * @dai: DAI
+ * @stream: STREAM
+ * @direction: Stream direction(Playback/Capture)
+ * SoundWire subsystem doesn't have a notion of direction and we reuse
+ * the ASoC stream direction to configure sink/source ports.
+ * Playback maps to source ports and Capture for sink ports.
+ *
+ * This should be invoked with NULL to clear the stream set previously.
+ * Returns 0 on success, a negative error code otherwise.
+ */
+static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
+				void *stream, int direction)
+{
+	if (dai->driver->ops->set_sdw_stream)
+		return dai->driver->ops->set_sdw_stream(dai, stream, direction);
+	else
+		return -ENOTSUPP;
+}
+
+#endif