571 lines
19 KiB
C
571 lines
19 KiB
C
/* SPDX-License-Identifier: GPL-2.0
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*
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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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* 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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/* Describes the possible PCM format */
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/*
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* use SND_SOC_DAI_FORMAT_xx as eash shift.
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* see
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* snd_soc_runtime_get_dai_fmt()
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*/
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#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT 0
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#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_I2S (1 << SND_SOC_DAI_FORMAT_I2S)
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#define SND_SOC_POSSIBLE_DAIFMT_RIGHT_J (1 << SND_SOC_DAI_FORMAT_RIGHT_J)
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#define SND_SOC_POSSIBLE_DAIFMT_LEFT_J (1 << SND_SOC_DAI_FORMAT_LEFT_J)
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#define SND_SOC_POSSIBLE_DAIFMT_DSP_A (1 << SND_SOC_DAI_FORMAT_DSP_A)
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#define SND_SOC_POSSIBLE_DAIFMT_DSP_B (1 << SND_SOC_DAI_FORMAT_DSP_B)
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#define SND_SOC_POSSIBLE_DAIFMT_AC97 (1 << SND_SOC_DAI_FORMAT_AC97)
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#define SND_SOC_POSSIBLE_DAIFMT_PDM (1 << SND_SOC_DAI_FORMAT_PDM)
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/*
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* DAI Clock gating.
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*
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* DAI bit clocks can 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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/* Describes the possible PCM format */
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/*
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* define GATED -> CONT. GATED will be selected if both are selected.
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* see
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* snd_soc_runtime_get_dai_fmt()
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*/
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#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT 16
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#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_GATED (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_CONT (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
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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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/* Describes the possible PCM format */
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#define SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT 32
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#define SND_SOC_POSSIBLE_DAIFMT_INV_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_NB_NF (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_NB_IF (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_IB_NF (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_IB_IF (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
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/*
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* DAI hardware clock providers/consumers
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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 provider then the interface is
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* clk and frame consumer.
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*/
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#define SND_SOC_DAIFMT_CBP_CFP (1 << 12) /* codec clk provider & frame provider */
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#define SND_SOC_DAIFMT_CBC_CFP (2 << 12) /* codec clk consumer & frame provider */
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#define SND_SOC_DAIFMT_CBP_CFC (3 << 12) /* codec clk provider & frame consumer */
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#define SND_SOC_DAIFMT_CBC_CFC (4 << 12) /* codec clk consumer & frame consumer */
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/* previous definitions kept for backwards-compatibility, do not use in new contributions */
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#define SND_SOC_DAIFMT_CBM_CFM SND_SOC_DAIFMT_CBP_CFP
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#define SND_SOC_DAIFMT_CBS_CFM SND_SOC_DAIFMT_CBC_CFP
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#define SND_SOC_DAIFMT_CBM_CFS SND_SOC_DAIFMT_CBP_CFC
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#define SND_SOC_DAIFMT_CBS_CFS SND_SOC_DAIFMT_CBC_CFC
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/* when passed to set_fmt directly indicate if the device is provider or consumer */
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#define SND_SOC_DAIFMT_BP_FP SND_SOC_DAIFMT_CBP_CFP
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#define SND_SOC_DAIFMT_BC_FP SND_SOC_DAIFMT_CBC_CFP
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#define SND_SOC_DAIFMT_BP_FC SND_SOC_DAIFMT_CBP_CFC
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#define SND_SOC_DAIFMT_BC_FC SND_SOC_DAIFMT_CBC_CFC
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/* Describes the possible PCM format */
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#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT 48
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#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFP (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFP (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFC (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
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#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFC (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
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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_CLOCK_PROVIDER_MASK 0xf000
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#define SND_SOC_DAIFMT_MASTER_MASK SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK
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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_S20_LE |\
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SNDRV_PCM_FMTBIT_S20_BE |\
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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_get_fmt_max_priority(struct snd_soc_pcm_runtime *rtd);
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u64 snd_soc_dai_get_fmt(struct snd_soc_dai *dai, int priority);
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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_get_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_is_dummy(struct snd_soc_dai *dai);
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int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params);
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void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream,
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int rollback);
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int snd_soc_dai_startup(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream);
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void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream, int rollback);
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void snd_soc_dai_suspend(struct snd_soc_dai *dai);
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void snd_soc_dai_resume(struct snd_soc_dai *dai);
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int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
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struct snd_soc_pcm_runtime *rtd, int num);
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bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
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void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
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void snd_soc_dai_action(struct snd_soc_dai *dai,
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int stream, int action);
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static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
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int stream)
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{
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snd_soc_dai_action(dai, stream, 1);
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}
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static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai,
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int stream)
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{
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snd_soc_dai_action(dai, stream, -1);
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}
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int snd_soc_dai_active(struct snd_soc_dai *dai);
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int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order);
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int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order);
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int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd);
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int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream);
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int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd,
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int rollback);
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int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream,
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int cmd);
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void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream,
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snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay);
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int snd_soc_dai_compr_startup(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream);
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void snd_soc_dai_compr_shutdown(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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int rollback);
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int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream, int cmd);
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int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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struct snd_compr_params *params);
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int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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struct snd_codec *params);
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int snd_soc_dai_compr_ack(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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size_t bytes);
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int snd_soc_dai_compr_pointer(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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struct snd_compr_tstamp *tstamp);
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int snd_soc_dai_compr_set_metadata(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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struct snd_compr_metadata *metadata);
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int snd_soc_dai_compr_get_metadata(struct snd_soc_dai *dai,
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struct snd_compr_stream *cstream,
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struct snd_compr_metadata *metadata);
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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 (*get_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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int (*set_stream)(struct snd_soc_dai *dai,
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void *stream, int direction);
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void *(*get_stream)(struct snd_soc_dai *dai, int direction);
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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 (*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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* Format list for auto selection.
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* Format will be increased if priority format was
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* not selected.
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* see
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* snd_soc_dai_get_fmt()
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*/
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u64 *auto_selectable_formats;
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int num_auto_selectable_formats;
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/* bit field */
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unsigned int no_capture_mute:1;
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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,
|
|
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);
|
|
/* 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);
|
|
|
|
/* 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_rate:1;
|
|
unsigned int symmetric_channels:1;
|
|
unsigned int symmetric_sample_bits: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 stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
|
|
|
|
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;
|
|
|
|
/* function mark */
|
|
struct snd_pcm_substream *mark_startup;
|
|
struct snd_pcm_substream *mark_hw_params;
|
|
struct snd_pcm_substream *mark_trigger;
|
|
struct snd_compr_stream *mark_compr_startup;
|
|
|
|
/* bit field */
|
|
unsigned int probed:1;
|
|
};
|
|
|
|
static inline struct snd_soc_pcm_stream *
|
|
snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
|
|
{
|
|
return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
|
|
&dai->driver->playback : &dai->driver->capture;
|
|
}
|
|
|
|
static inline
|
|
struct snd_soc_dapm_widget *snd_soc_dai_get_widget(
|
|
struct snd_soc_dai *dai, int stream)
|
|
{
|
|
return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
|
|
dai->playback_widget : dai->capture_widget;
|
|
}
|
|
|
|
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_stream() - Configures a DAI for stream operation
|
|
* @dai: DAI
|
|
* @stream: STREAM (opaque structure depending on DAI type)
|
|
* @direction: Stream direction(Playback/Capture)
|
|
* Some subsystems, such as SoundWire, don'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_stream(struct snd_soc_dai *dai,
|
|
void *stream, int direction)
|
|
{
|
|
if (dai->driver->ops->set_stream)
|
|
return dai->driver->ops->set_stream(dai, stream, direction);
|
|
else
|
|
return -ENOTSUPP;
|
|
}
|
|
|
|
/**
|
|
* snd_soc_dai_get_stream() - Retrieves stream from DAI
|
|
* @dai: DAI
|
|
* @direction: Stream direction(Playback/Capture)
|
|
*
|
|
* This routine only retrieves that was previously configured
|
|
* with snd_soc_dai_get_stream()
|
|
*
|
|
* Returns pointer to stream or an ERR_PTR value, e.g.
|
|
* ERR_PTR(-ENOTSUPP) if callback is not supported;
|
|
*/
|
|
static inline void *snd_soc_dai_get_stream(struct snd_soc_dai *dai,
|
|
int direction)
|
|
{
|
|
if (dai->driver->ops->get_stream)
|
|
return dai->driver->ops->get_stream(dai, direction);
|
|
else
|
|
return ERR_PTR(-ENOTSUPP);
|
|
}
|
|
|
|
static inline unsigned int
|
|
snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream)
|
|
{
|
|
return dai->stream_active[stream];
|
|
}
|
|
|
|
#endif
|