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