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![Chen Zhongjin](/assets/img/avatar_default.png)
snd_soc_util_exit() is called in __init snd_soc_init() for cleanup.
Remove the __exit annotation for it to fix the build warning:
WARNING: modpost: sound/soc/snd-soc-core.o: section mismatch in reference: init_module (section: .init.text) -> snd_soc_util_exit (section: .exit.text)
Fixes: 6ec27c5388
("ASoC: core: Fix use-after-free in snd_soc_exit()")
Signed-off-by: Chen Zhongjin <chenzhongjin@huawei.com>
Link: https://lore.kernel.org/r/20221031134031.256511-1-chenzhongjin@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
271 lines
7.1 KiB
C
271 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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#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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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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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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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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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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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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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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ret = snd_soc_params_to_frame_size(params);
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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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* 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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if (slot_multiple > 1)
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tdm_slots = roundup(tdm_slots, slot_multiple);
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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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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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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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static const struct snd_soc_component_driver dummy_platform;
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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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* 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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/* 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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return 0;
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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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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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#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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* 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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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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* 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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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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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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static int snd_soc_dummy_probe(struct platform_device *pdev)
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{
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int ret;
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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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ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform,
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NULL, 0);
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return ret;
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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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static struct platform_device *soc_dummy_dev;
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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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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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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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return ret;
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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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