acf253c113
DAI driver has playback/capture stream. OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Because of this kind of implementation, ALSA SoC needs to have many verbose code. To solve this issue, this patch adds snd_soc_dai_get_pcm_stream() macro to get playback/capture stream pointer from stream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87ftf7jcab.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
419 lines
13 KiB
C
419 lines
13 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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/*
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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_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_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 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);
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int snd_soc_dai_prepare(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream);
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int snd_soc_dai_trigger(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream, int cmd);
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int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream, int cmd);
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snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
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struct snd_pcm_substream *substream);
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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_probe(struct snd_soc_dai *dai);
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int snd_soc_dai_remove(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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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_sdw_stream)(struct snd_soc_dai *dai,
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void *stream, 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 (*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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/* 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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/* 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 stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
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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_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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/* bit field */
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unsigned int probed:1;
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unsigned int started:1;
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};
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static inline struct snd_soc_pcm_stream *
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snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
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{
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return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
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&dai->driver->playback : &dai->driver->capture;
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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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/**
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* snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
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* @dai: DAI
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* @stream: STREAM
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* @direction: Stream direction(Playback/Capture)
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* SoundWire subsystem doesn't have a notion of direction and we reuse
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* the ASoC stream direction to configure sink/source ports.
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* Playback maps to source ports and Capture for sink ports.
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*
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* This should be invoked with NULL to clear the stream set previously.
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* Returns 0 on success, a negative error code otherwise.
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*/
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static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
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void *stream, int direction)
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{
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if (dai->driver->ops->set_sdw_stream)
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return dai->driver->ops->set_sdw_stream(dai, stream, direction);
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else
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return -ENOTSUPP;
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}
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#endif
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