25f7b701c2
During probe, DAIs can need to perform some actions that requests the knowledge of the pcm runtime handle. The callback is called during DAIs linking, after PCM device creation. For instance this can be used to add relationship between a DAI pcm control and the pcm device. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
363 lines
11 KiB
C
363 lines
11 KiB
C
/*
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* linux/sound/soc-dai.h -- ALSA SoC Layer
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*
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* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 as
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* published by the Free Software Foundation.
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*
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* Digital Audio Interface (DAI) API.
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*/
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#ifndef __LINUX_SND_SOC_DAI_H
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#define __LINUX_SND_SOC_DAI_H
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#include <linux/list.h>
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#include <sound/asoc.h>
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struct snd_pcm_substream;
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struct snd_soc_dapm_widget;
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struct snd_compr_stream;
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/*
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* DAI hardware audio formats.
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*
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* Describes the physical PCM data formating and clocking. Add new formats
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* to the end.
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*/
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#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
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#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
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#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
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#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
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#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
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#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
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#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
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/* left and right justified also known as MSB and LSB respectively */
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#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
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#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
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/*
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* DAI Clock gating.
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*
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* DAI bit clocks can be be gated (disabled) when the DAI is not
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* sending or receiving PCM data in a frame. This can be used to save power.
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*/
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#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
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#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
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/*
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* DAI hardware signal polarity.
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*
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* Specifies whether the DAI can also support inverted clocks for the specified
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* format.
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*
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* BCLK:
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* - "normal" polarity means signal is available at rising edge of BCLK
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* - "inverted" polarity means signal is available at falling edge of BCLK
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*
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* FSYNC "normal" polarity depends on the frame format:
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* - I2S: frame consists of left then right channel data. Left channel starts
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* with falling FSYNC edge, right channel starts with rising FSYNC edge.
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* - Left/Right Justified: frame consists of left then right channel data.
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* Left channel starts with rising FSYNC edge, right channel starts with
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* falling FSYNC edge.
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* - DSP A/B: Frame starts with rising FSYNC edge.
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* - AC97: Frame starts with rising FSYNC edge.
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*
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* "Negative" FSYNC polarity is the one opposite of "normal" polarity.
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*/
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#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
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#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
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#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
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/*
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* DAI hardware clock masters.
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*
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* This is wrt the codec, the inverse is true for the interface
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* i.e. if the codec is clk and FRM master then the interface is
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* clk and frame slave.
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*/
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#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
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#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
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#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
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#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
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#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
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#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
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#define SND_SOC_DAIFMT_INV_MASK 0x0f00
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#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
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/*
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* Master Clock Directions
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*/
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#define SND_SOC_CLOCK_IN 0
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#define SND_SOC_CLOCK_OUT 1
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#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
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SNDRV_PCM_FMTBIT_S16_LE |\
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SNDRV_PCM_FMTBIT_S16_BE |\
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SNDRV_PCM_FMTBIT_S20_3LE |\
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SNDRV_PCM_FMTBIT_S20_3BE |\
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SNDRV_PCM_FMTBIT_S24_3LE |\
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SNDRV_PCM_FMTBIT_S24_3BE |\
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SNDRV_PCM_FMTBIT_S32_LE |\
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SNDRV_PCM_FMTBIT_S32_BE)
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struct snd_soc_dai_driver;
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struct snd_soc_dai;
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struct snd_ac97_bus_ops;
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/* Digital Audio Interface clocking API.*/
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int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
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unsigned int freq, int dir);
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int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
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int div_id, int div);
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int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
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int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
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int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
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/* Digital Audio interface formatting */
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int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
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int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
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unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
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int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
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unsigned int tx_num, unsigned int *tx_slot,
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unsigned int rx_num, unsigned int *rx_slot);
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int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
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/* Digital Audio Interface mute */
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int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
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int direction);
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int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
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struct snd_soc_dai_ops {
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/*
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* DAI clocking configuration, all optional.
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* Called by soc_card drivers, normally in their hw_params.
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*/
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int (*set_sysclk)(struct snd_soc_dai *dai,
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int clk_id, unsigned int freq, int dir);
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int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
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unsigned int freq_in, unsigned int freq_out);
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int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
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int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
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/*
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* DAI format configuration
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* Called by soc_card drivers, normally in their hw_params.
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*/
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int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
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int (*xlate_tdm_slot_mask)(unsigned int slots,
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unsigned int *tx_mask, unsigned int *rx_mask);
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int (*set_tdm_slot)(struct snd_soc_dai *dai,
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unsigned int tx_mask, unsigned int rx_mask,
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int slots, int slot_width);
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int (*set_channel_map)(struct snd_soc_dai *dai,
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unsigned int tx_num, unsigned int *tx_slot,
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unsigned int rx_num, unsigned int *rx_slot);
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int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
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/*
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* DAI digital mute - optional.
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* Called by soc-core to minimise any pops.
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*/
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int (*digital_mute)(struct snd_soc_dai *dai, int mute);
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int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
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/*
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* ALSA PCM audio operations - all optional.
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* Called by soc-core during audio PCM operations.
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*/
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int (*startup)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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void (*shutdown)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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int (*hw_params)(struct snd_pcm_substream *,
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struct snd_pcm_hw_params *, struct snd_soc_dai *);
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int (*hw_free)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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int (*prepare)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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/*
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* NOTE: Commands passed to the trigger function are not necessarily
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* compatible with the current state of the dai. For example this
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* sequence of commands is possible: START STOP STOP.
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* So do not unconditionally use refcounting functions in the trigger
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* function, e.g. clk_enable/disable.
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*/
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int (*trigger)(struct snd_pcm_substream *, int,
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struct snd_soc_dai *);
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int (*bespoke_trigger)(struct snd_pcm_substream *, int,
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struct snd_soc_dai *);
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/*
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* For hardware based FIFO caused delay reporting.
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* Optional.
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*/
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snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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};
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struct snd_soc_cdai_ops {
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/*
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* for compress ops
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*/
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int (*startup)(struct snd_compr_stream *,
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struct snd_soc_dai *);
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int (*shutdown)(struct snd_compr_stream *,
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struct snd_soc_dai *);
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int (*set_params)(struct snd_compr_stream *,
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struct snd_compr_params *, struct snd_soc_dai *);
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int (*get_params)(struct snd_compr_stream *,
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struct snd_codec *, struct snd_soc_dai *);
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int (*set_metadata)(struct snd_compr_stream *,
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struct snd_compr_metadata *, struct snd_soc_dai *);
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int (*get_metadata)(struct snd_compr_stream *,
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struct snd_compr_metadata *, struct snd_soc_dai *);
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int (*trigger)(struct snd_compr_stream *, int,
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struct snd_soc_dai *);
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int (*pointer)(struct snd_compr_stream *,
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struct snd_compr_tstamp *, struct snd_soc_dai *);
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int (*ack)(struct snd_compr_stream *, size_t,
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struct snd_soc_dai *);
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};
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/*
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* Digital Audio Interface Driver.
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*
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* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
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* operations and capabilities. Codec and platform drivers will register this
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* structure for every DAI they have.
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*
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* This structure covers the clocking, formating and ALSA operations for each
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* interface.
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*/
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struct snd_soc_dai_driver {
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/* DAI description */
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const char *name;
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unsigned int id;
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unsigned int base;
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struct snd_soc_dobj dobj;
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/* DAI driver callbacks */
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int (*probe)(struct snd_soc_dai *dai);
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int (*remove)(struct snd_soc_dai *dai);
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int (*suspend)(struct snd_soc_dai *dai);
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int (*resume)(struct snd_soc_dai *dai);
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/* compress dai */
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int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
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/* Optional Callback used at pcm creation*/
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int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
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struct snd_soc_dai *dai);
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/* DAI is also used for the control bus */
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bool bus_control;
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/* ops */
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const struct snd_soc_dai_ops *ops;
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const struct snd_soc_cdai_ops *cops;
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/* DAI capabilities */
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struct snd_soc_pcm_stream capture;
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struct snd_soc_pcm_stream playback;
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unsigned int symmetric_rates:1;
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unsigned int symmetric_channels:1;
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unsigned int symmetric_samplebits:1;
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/* probe ordering - for components with runtime dependencies */
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int probe_order;
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int remove_order;
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};
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/*
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* Digital Audio Interface runtime data.
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*
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* Holds runtime data for a DAI.
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*/
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struct snd_soc_dai {
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const char *name;
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int id;
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struct device *dev;
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/* driver ops */
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struct snd_soc_dai_driver *driver;
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/* DAI runtime info */
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unsigned int capture_active:1; /* stream is in use */
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unsigned int playback_active:1; /* stream is in use */
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unsigned int symmetric_rates:1;
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unsigned int symmetric_channels:1;
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unsigned int symmetric_samplebits:1;
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unsigned int probed:1;
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unsigned int active;
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struct snd_soc_dapm_widget *playback_widget;
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struct snd_soc_dapm_widget *capture_widget;
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/* DAI DMA data */
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void *playback_dma_data;
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void *capture_dma_data;
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/* Symmetry data - only valid if symmetry is being enforced */
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unsigned int rate;
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unsigned int channels;
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unsigned int sample_bits;
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/* parent platform/codec */
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struct snd_soc_codec *codec;
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struct snd_soc_component *component;
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/* CODEC TDM slot masks and params (for fixup) */
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unsigned int tx_mask;
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unsigned int rx_mask;
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struct list_head list;
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};
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static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
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const struct snd_pcm_substream *ss)
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{
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return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
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dai->playback_dma_data : dai->capture_dma_data;
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}
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static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
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const struct snd_pcm_substream *ss,
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void *data)
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{
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if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
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dai->playback_dma_data = data;
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else
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dai->capture_dma_data = data;
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}
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static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
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void *playback, void *capture)
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{
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dai->playback_dma_data = playback;
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dai->capture_dma_data = capture;
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}
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static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
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void *data)
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{
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dev_set_drvdata(dai->dev, data);
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}
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static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
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{
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return dev_get_drvdata(dai->dev);
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}
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#endif
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