// SPDX-License-Identifier: GPL-2.0+
//
// soc-util.c  --  ALSA SoC Audio Layer utility functions
//
// Copyright 2009 Wolfson Microelectronics PLC.
//
// Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
//         Liam Girdwood <lrg@slimlogic.co.uk>

#include <linux/device/faux.h>
#include <linux/export.h>
#include <linux/math.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>

int snd_soc_ret(const struct device *dev, int ret, const char *fmt, ...)
{
        struct va_format vaf;
        va_list args;

        /* Positive, Zero values are not errors */
        if (ret >= 0)
                return ret;

        /* Negative values might be errors */
        switch (ret) {
        case -EPROBE_DEFER:
        case -ENOTSUPP:
        case -EOPNOTSUPP:
                break;
        default:
                va_start(args, fmt);
                vaf.fmt = fmt;
                vaf.va = &args;

                dev_err(dev, "ASoC error (%d): %pV", ret, &vaf);
        }

        return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_ret);

int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots)
{
        return sample_size * channels * tdm_slots;
}
EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size);

int snd_soc_params_to_frame_size(const struct snd_pcm_hw_params *params)
{
        int sample_size;

        sample_size = snd_pcm_format_width(params_format(params));
        if (sample_size < 0)
                return sample_size;

        return snd_soc_calc_frame_size(sample_size, params_channels(params),
                                       1);
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size);

int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
{
        return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots);
}
EXPORT_SYMBOL_GPL(snd_soc_calc_bclk);

int snd_soc_params_to_bclk(const struct snd_pcm_hw_params *params)
{
        int ret;

        ret = snd_soc_params_to_frame_size(params);

        if (ret > 0)
                return ret * params_rate(params);
        else
                return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);

/**
 * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info.
 *
 * Calculate the bclk from the params sample rate, the tdm slot count and the
 * tdm slot width. Optionally round-up the slot count to a given multiple.
 * Either or both of tdm_width and tdm_slots can be 0.
 *
 * If tdm_width == 0:   use params_width() as the slot width.
 * If tdm_slots == 0:   use params_channels() as the slot count.
 *
 * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0)
 * will be rounded up to a multiple of slot_multiple. This is mainly useful for
 * I2S mode, which has a left and right phase so the number of slots is always
 * a multiple of 2.
 *
 * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent
 * to calling snd_soc_params_to_bclk().
 *
 * @params:        Pointer to struct_pcm_hw_params.
 * @tdm_width:     Width in bits of the tdm slots. Must be >= 0.
 * @tdm_slots:     Number of tdm slots per frame. Must be >= 0.
 * @slot_multiple: If >1 roundup slot count to a multiple of this value.
 *
 * Return: bclk frequency in Hz, else a negative error code if params format
 *         is invalid.
 */
int snd_soc_tdm_params_to_bclk(const struct snd_pcm_hw_params *params,
                               int tdm_width, int tdm_slots, int slot_multiple)
{
        if (!tdm_slots)
                tdm_slots = params_channels(params);

        if (slot_multiple > 1)
                tdm_slots = roundup(tdm_slots, slot_multiple);

        if (!tdm_width) {
                tdm_width = snd_pcm_format_width(params_format(params));
                if (tdm_width < 0)
                        return tdm_width;
        }

        return snd_soc_calc_bclk(params_rate(params), tdm_width, 1, tdm_slots);
}
EXPORT_SYMBOL_GPL(snd_soc_tdm_params_to_bclk);

static const struct snd_pcm_hardware dummy_dma_hardware = {
        /* Random values to keep userspace happy when checking constraints */
        .info                   = SNDRV_PCM_INFO_INTERLEAVED |
                                  SNDRV_PCM_INFO_BLOCK_TRANSFER,
        .buffer_bytes_max       = 128*1024,
        .period_bytes_min       = 4096,
        .period_bytes_max       = 4096*2,
        .periods_min            = 2,
        .periods_max            = 128,
};


static const struct snd_soc_component_driver dummy_platform;

static int dummy_dma_open(struct snd_soc_component *component,
                          struct snd_pcm_substream *substream)
{
        struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
        int i;

        /*
         * If there are other components associated with rtd, we shouldn't
         * override their hwparams
         */
        for_each_rtd_components(rtd, i, component) {
                if (component->driver == &dummy_platform)
                        return 0;
        }

        /* BE's dont need dummy params */
        if (!rtd->dai_link->no_pcm)
                snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);

        return 0;
}

static const struct snd_soc_component_driver dummy_platform = {
        .open           = dummy_dma_open,
};

static const struct snd_soc_component_driver dummy_codec = {
        .idle_bias_on           = 1,
        .use_pmdown_time        = 1,
        .endianness             = 1,
};

#define STUB_FORMATS    (SNDRV_PCM_FMTBIT_S8 | \
                        SNDRV_PCM_FMTBIT_U8 | \
                        SNDRV_PCM_FMTBIT_S16_LE | \
                        SNDRV_PCM_FMTBIT_U16_LE | \
                        SNDRV_PCM_FMTBIT_S24_LE | \
                        SNDRV_PCM_FMTBIT_S24_3LE | \
                        SNDRV_PCM_FMTBIT_U24_LE | \
                        SNDRV_PCM_FMTBIT_S32_LE | \
                        SNDRV_PCM_FMTBIT_U32_LE | \
                        SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)

/*
 * Select these from Sound Card Manually
 *      SND_SOC_POSSIBLE_DAIFMT_CBP_CFP
 *      SND_SOC_POSSIBLE_DAIFMT_CBP_CFC
 *      SND_SOC_POSSIBLE_DAIFMT_CBC_CFP
 *      SND_SOC_POSSIBLE_DAIFMT_CBC_CFC
 */
static const u64 dummy_dai_formats =
        SND_SOC_POSSIBLE_DAIFMT_I2S     |
        SND_SOC_POSSIBLE_DAIFMT_RIGHT_J |
        SND_SOC_POSSIBLE_DAIFMT_LEFT_J  |
        SND_SOC_POSSIBLE_DAIFMT_DSP_A   |
        SND_SOC_POSSIBLE_DAIFMT_DSP_B   |
        SND_SOC_POSSIBLE_DAIFMT_AC97    |
        SND_SOC_POSSIBLE_DAIFMT_PDM     |
        SND_SOC_POSSIBLE_DAIFMT_GATED   |
        SND_SOC_POSSIBLE_DAIFMT_CONT    |
        SND_SOC_POSSIBLE_DAIFMT_NB_NF   |
        SND_SOC_POSSIBLE_DAIFMT_NB_IF   |
        SND_SOC_POSSIBLE_DAIFMT_IB_NF   |
        SND_SOC_POSSIBLE_DAIFMT_IB_IF;

static const struct snd_soc_dai_ops dummy_dai_ops = {
        .auto_selectable_formats        = &dummy_dai_formats,
        .num_auto_selectable_formats    = 1,
};

/*
 * The dummy CODEC is only meant to be used in situations where there is no
 * actual hardware.
 *
 * If there is actual hardware even if it does not have a control bus
 * the hardware will still have constraints like supported samplerates, etc.
 * which should be modelled. And the data flow graph also should be modelled
 * using DAPM.
 */
static struct snd_soc_dai_driver dummy_dai = {
        .name = "snd-soc-dummy-dai",
        .playback = {
                .stream_name    = "Playback",
                .channels_min   = 1,
                .channels_max   = 384,
                .rates          = SNDRV_PCM_RATE_CONTINUOUS,
                .rate_min       = 5512,
                .rate_max       = 768000,
                .formats        = STUB_FORMATS,
        },
        .capture = {
                .stream_name    = "Capture",
                .channels_min   = 1,
                .channels_max   = 384,
                .rates = SNDRV_PCM_RATE_CONTINUOUS,
                .rate_min       = 5512,
                .rate_max       = 768000,
                .formats = STUB_FORMATS,
         },
        .ops = &dummy_dai_ops,
};

int snd_soc_dai_is_dummy(const struct snd_soc_dai *dai)
{
        if (dai->driver == &dummy_dai)
                return 1;
        return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy);

int snd_soc_component_is_dummy(struct snd_soc_component *component)
{
        return ((component->driver == &dummy_platform) ||
                (component->driver == &dummy_codec));
}

struct snd_soc_dai_link_component snd_soc_dummy_dlc = {
        .of_node        = NULL,
        .dai_name       = "snd-soc-dummy-dai",
        .name           = "snd-soc-dummy",
};
EXPORT_SYMBOL_GPL(snd_soc_dummy_dlc);

int snd_soc_dlc_is_dummy(struct snd_soc_dai_link_component *dlc)
{
        if (dlc == &snd_soc_dummy_dlc)
                return true;

        if ((dlc->name     && strcmp(dlc->name,     snd_soc_dummy_dlc.name)     == 0) ||
            (dlc->dai_name && strcmp(dlc->dai_name, snd_soc_dummy_dlc.dai_name) == 0))
                return true;

        return false;
}
EXPORT_SYMBOL_GPL(snd_soc_dlc_is_dummy);

static int snd_soc_dummy_probe(struct faux_device *fdev)
{
        int ret;

        ret = devm_snd_soc_register_component(&fdev->dev,
                                              &dummy_codec, &dummy_dai, 1);
        if (ret < 0)
                return ret;

        ret = devm_snd_soc_register_component(&fdev->dev, &dummy_platform,
                                              NULL, 0);

        return ret;
}

static struct faux_device_ops soc_dummy_ops = {
        .probe = snd_soc_dummy_probe,
};

static struct faux_device *soc_dummy_dev;

int __init snd_soc_util_init(void)
{
        soc_dummy_dev = faux_device_create("snd-soc-dummy", NULL,
                                           &soc_dummy_ops);
        if (!soc_dummy_dev)
                return -ENODEV;

        return 0;
}

void snd_soc_util_exit(void)
{
        faux_device_destroy(soc_dummy_dev);
}