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![Julia Lawall](/assets/img/avatar_default.png)
Check for snd_soc_ops structures that are only stored in the ops field of a snd_soc_dai_link structure. This field is declared const, so snd_soc_ops structures that have this property can be declared as const also. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r disable optional_qualifier@ identifier i; position p; @@ static struct snd_soc_ops i@p = { ... }; @ok1@ identifier r.i; struct snd_soc_dai_link e; position p; @@ e.ops = &i@p; @ok2@ identifier r.i, e; position p; @@ struct snd_soc_dai_link e[] = { ..., { .ops = &i@p, }, ..., }; @bad@ position p != {r.p,ok1.p,ok2.p}; identifier r.i; struct snd_soc_ops e; @@ e@i@p @depends on !bad disable optional_qualifier@ identifier r.i; @@ static +const struct snd_soc_ops i = { ... }; // </smpl> The effect on the layout of the .o files is shown by the following output of the size command, first before then after the transformation: text data bss dec hex filename 8748 1024 0 9772 262c sound/soc/fsl/fsl-asoc-card.o 8812 952 0 9764 2624 sound/soc/fsl/fsl-asoc-card.o text data bss dec hex filename 4165 264 8 4437 1155 sound/soc/fsl/imx-wm8962.o 4229 200 8 4437 1155 sound/soc/fsl/imx-wm8962.o Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
722 lines
20 KiB
C
722 lines
20 KiB
C
/*
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* Freescale Generic ASoC Sound Card driver with ASRC
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*
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* Copyright (C) 2014 Freescale Semiconductor, Inc.
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*
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* Author: Nicolin Chen <nicoleotsuka@gmail.com>
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*
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* This file is licensed under the terms of the GNU General Public License
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* version 2. This program is licensed "as is" without any warranty of any
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* kind, whether express or implied.
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*/
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#include <linux/clk.h>
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#include <linux/i2c.h>
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#include <linux/module.h>
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#include <linux/of_platform.h>
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#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
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#include <sound/ac97_codec.h>
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#endif
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include "fsl_esai.h"
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#include "fsl_sai.h"
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#include "imx-audmux.h"
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#include "../codecs/sgtl5000.h"
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#include "../codecs/wm8962.h"
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#include "../codecs/wm8960.h"
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#define CS427x_SYSCLK_MCLK 0
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#define RX 0
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#define TX 1
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/* Default DAI format without Master and Slave flag */
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#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
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/**
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* CODEC private data
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*
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* @mclk_freq: Clock rate of MCLK
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* @mclk_id: MCLK (or main clock) id for set_sysclk()
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* @fll_id: FLL (or secordary clock) id for set_sysclk()
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* @pll_id: PLL id for set_pll()
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*/
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struct codec_priv {
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unsigned long mclk_freq;
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u32 mclk_id;
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u32 fll_id;
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u32 pll_id;
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};
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/**
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* CPU private data
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*
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* @sysclk_freq[2]: SYSCLK rates for set_sysclk()
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* @sysclk_dir[2]: SYSCLK directions for set_sysclk()
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* @sysclk_id[2]: SYSCLK ids for set_sysclk()
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* @slot_width: Slot width of each frame
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*
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* Note: [1] for tx and [0] for rx
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*/
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struct cpu_priv {
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unsigned long sysclk_freq[2];
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u32 sysclk_dir[2];
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u32 sysclk_id[2];
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u32 slot_width;
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};
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/**
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* Freescale Generic ASOC card private data
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*
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* @dai_link[3]: DAI link structure including normal one and DPCM link
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* @pdev: platform device pointer
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* @codec_priv: CODEC private data
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* @cpu_priv: CPU private data
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* @card: ASoC card structure
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* @sample_rate: Current sample rate
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* @sample_format: Current sample format
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* @asrc_rate: ASRC sample rate used by Back-Ends
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* @asrc_format: ASRC sample format used by Back-Ends
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* @dai_fmt: DAI format between CPU and CODEC
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* @name: Card name
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*/
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struct fsl_asoc_card_priv {
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struct snd_soc_dai_link dai_link[3];
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struct platform_device *pdev;
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struct codec_priv codec_priv;
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struct cpu_priv cpu_priv;
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struct snd_soc_card card;
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u32 sample_rate;
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u32 sample_format;
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u32 asrc_rate;
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u32 asrc_format;
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u32 dai_fmt;
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char name[32];
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};
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/**
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* This dapm route map exsits for DPCM link only.
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* The other routes shall go through Device Tree.
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*
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* Note: keep all ASRC routes in the second half
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* to drop them easily for non-ASRC cases.
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*/
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static const struct snd_soc_dapm_route audio_map[] = {
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/* 1st half -- Normal DAPM routes */
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{"Playback", NULL, "CPU-Playback"},
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{"CPU-Capture", NULL, "Capture"},
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/* 2nd half -- ASRC DAPM routes */
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{"CPU-Playback", NULL, "ASRC-Playback"},
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{"ASRC-Capture", NULL, "CPU-Capture"},
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};
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static const struct snd_soc_dapm_route audio_map_ac97[] = {
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/* 1st half -- Normal DAPM routes */
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{"Playback", NULL, "AC97 Playback"},
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{"AC97 Capture", NULL, "Capture"},
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/* 2nd half -- ASRC DAPM routes */
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{"AC97 Playback", NULL, "ASRC-Playback"},
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{"ASRC-Capture", NULL, "AC97 Capture"},
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};
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/* Add all possible widgets into here without being redundant */
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static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
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SND_SOC_DAPM_LINE("Line Out Jack", NULL),
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SND_SOC_DAPM_LINE("Line In Jack", NULL),
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_MIC("Mic Jack", NULL),
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SND_SOC_DAPM_MIC("AMIC", NULL),
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SND_SOC_DAPM_MIC("DMIC", NULL),
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};
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static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
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{
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return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
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}
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static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
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bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
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struct cpu_priv *cpu_priv = &priv->cpu_priv;
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struct device *dev = rtd->card->dev;
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int ret;
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priv->sample_rate = params_rate(params);
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priv->sample_format = params_format(params);
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/*
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* If codec-dai is DAI Master and all configurations are already in the
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* set_bias_level(), bypass the remaining settings in hw_params().
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* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
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*/
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if ((priv->card.set_bias_level &&
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priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
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fsl_asoc_card_is_ac97(priv))
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return 0;
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/* Specific configurations of DAIs starts from here */
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ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
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cpu_priv->sysclk_freq[tx],
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cpu_priv->sysclk_dir[tx]);
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if (ret) {
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dev_err(dev, "failed to set sysclk for cpu dai\n");
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return ret;
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}
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if (cpu_priv->slot_width) {
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ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
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cpu_priv->slot_width);
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if (ret) {
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dev_err(dev, "failed to set TDM slot for cpu dai\n");
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return ret;
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}
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}
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return 0;
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}
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static const struct snd_soc_ops fsl_asoc_card_ops = {
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.hw_params = fsl_asoc_card_hw_params,
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};
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static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
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struct snd_pcm_hw_params *params)
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{
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struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
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struct snd_interval *rate;
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struct snd_mask *mask;
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rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
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rate->max = rate->min = priv->asrc_rate;
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mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
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snd_mask_none(mask);
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snd_mask_set(mask, priv->asrc_format);
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return 0;
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}
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static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
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/* Default ASoC DAI Link*/
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{
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.name = "HiFi",
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.stream_name = "HiFi",
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.ops = &fsl_asoc_card_ops,
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},
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/* DPCM Link between Front-End and Back-End (Optional) */
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{
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.name = "HiFi-ASRC-FE",
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.stream_name = "HiFi-ASRC-FE",
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.codec_name = "snd-soc-dummy",
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.codec_dai_name = "snd-soc-dummy-dai",
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.dynamic = 1,
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},
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{
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.name = "HiFi-ASRC-BE",
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.stream_name = "HiFi-ASRC-BE",
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.platform_name = "snd-soc-dummy",
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.be_hw_params_fixup = be_hw_params_fixup,
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.ops = &fsl_asoc_card_ops,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.no_pcm = 1,
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},
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};
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static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
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struct snd_soc_dapm_context *dapm,
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enum snd_soc_bias_level level)
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{
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struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
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struct snd_soc_pcm_runtime *rtd;
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struct snd_soc_dai *codec_dai;
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struct codec_priv *codec_priv = &priv->codec_priv;
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struct device *dev = card->dev;
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unsigned int pll_out;
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int ret;
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rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
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codec_dai = rtd->codec_dai;
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if (dapm->dev != codec_dai->dev)
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return 0;
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switch (level) {
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case SND_SOC_BIAS_PREPARE:
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if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
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break;
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if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
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pll_out = priv->sample_rate * 384;
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else
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pll_out = priv->sample_rate * 256;
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ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
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codec_priv->mclk_id,
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codec_priv->mclk_freq, pll_out);
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if (ret) {
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dev_err(dev, "failed to start FLL: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
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pll_out, SND_SOC_CLOCK_IN);
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if (ret) {
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dev_err(dev, "failed to set SYSCLK: %d\n", ret);
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return ret;
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}
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break;
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case SND_SOC_BIAS_STANDBY:
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if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
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break;
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ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
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codec_priv->mclk_freq,
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SND_SOC_CLOCK_IN);
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if (ret) {
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dev_err(dev, "failed to switch away from FLL: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
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if (ret) {
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dev_err(dev, "failed to stop FLL: %d\n", ret);
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return ret;
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}
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break;
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default:
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break;
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}
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return 0;
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}
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static int fsl_asoc_card_audmux_init(struct device_node *np,
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struct fsl_asoc_card_priv *priv)
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{
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struct device *dev = &priv->pdev->dev;
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u32 int_ptcr = 0, ext_ptcr = 0;
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int int_port, ext_port;
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int ret;
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ret = of_property_read_u32(np, "mux-int-port", &int_port);
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if (ret) {
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dev_err(dev, "mux-int-port missing or invalid\n");
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return ret;
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}
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ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
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if (ret) {
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dev_err(dev, "mux-ext-port missing or invalid\n");
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return ret;
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}
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/*
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* The port numbering in the hardware manual starts at 1, while
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* the AUDMUX API expects it starts at 0.
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*/
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int_port--;
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ext_port--;
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/*
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* Use asynchronous mode (6 wires) for all cases except AC97.
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* If only 4 wires are needed, just set SSI into
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* synchronous mode and enable 4 PADs in IOMUX.
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*/
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switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
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case SND_SOC_DAIFMT_CBM_CFM:
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int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
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IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
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IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
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IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
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IMX_AUDMUX_V2_PTCR_RFSDIR |
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IMX_AUDMUX_V2_PTCR_RCLKDIR |
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IMX_AUDMUX_V2_PTCR_TFSDIR |
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IMX_AUDMUX_V2_PTCR_TCLKDIR;
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break;
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case SND_SOC_DAIFMT_CBM_CFS:
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int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
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IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
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IMX_AUDMUX_V2_PTCR_RCLKDIR |
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IMX_AUDMUX_V2_PTCR_TCLKDIR;
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ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
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IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
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IMX_AUDMUX_V2_PTCR_RFSDIR |
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IMX_AUDMUX_V2_PTCR_TFSDIR;
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break;
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case SND_SOC_DAIFMT_CBS_CFM:
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int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
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IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
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IMX_AUDMUX_V2_PTCR_RFSDIR |
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IMX_AUDMUX_V2_PTCR_TFSDIR;
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ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
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IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
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IMX_AUDMUX_V2_PTCR_RCLKDIR |
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IMX_AUDMUX_V2_PTCR_TCLKDIR;
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break;
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case SND_SOC_DAIFMT_CBS_CFS:
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ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
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IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
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IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
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IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
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IMX_AUDMUX_V2_PTCR_RFSDIR |
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IMX_AUDMUX_V2_PTCR_RCLKDIR |
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IMX_AUDMUX_V2_PTCR_TFSDIR |
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IMX_AUDMUX_V2_PTCR_TCLKDIR;
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break;
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default:
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if (!fsl_asoc_card_is_ac97(priv))
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return -EINVAL;
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}
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if (fsl_asoc_card_is_ac97(priv)) {
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int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
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IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
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IMX_AUDMUX_V2_PTCR_TCLKDIR;
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ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
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IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
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IMX_AUDMUX_V2_PTCR_TFSDIR;
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}
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/* Asynchronous mode can not be set along with RCLKDIR */
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if (!fsl_asoc_card_is_ac97(priv)) {
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unsigned int pdcr =
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IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
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ret = imx_audmux_v2_configure_port(int_port, 0,
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pdcr);
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if (ret) {
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dev_err(dev, "audmux internal port setup failed\n");
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return ret;
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}
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}
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ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
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IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
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if (ret) {
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dev_err(dev, "audmux internal port setup failed\n");
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return ret;
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}
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if (!fsl_asoc_card_is_ac97(priv)) {
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unsigned int pdcr =
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IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
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ret = imx_audmux_v2_configure_port(ext_port, 0,
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pdcr);
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if (ret) {
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dev_err(dev, "audmux external port setup failed\n");
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return ret;
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}
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}
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ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
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IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
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if (ret) {
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dev_err(dev, "audmux external port setup failed\n");
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return ret;
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}
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return 0;
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}
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static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
|
|
{
|
|
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
|
|
struct snd_soc_pcm_runtime *rtd = list_first_entry(
|
|
&card->rtd_list, struct snd_soc_pcm_runtime, list);
|
|
struct snd_soc_dai *codec_dai = rtd->codec_dai;
|
|
struct codec_priv *codec_priv = &priv->codec_priv;
|
|
struct device *dev = card->dev;
|
|
int ret;
|
|
|
|
if (fsl_asoc_card_is_ac97(priv)) {
|
|
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
|
|
struct snd_soc_codec *codec = rtd->codec;
|
|
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
|
|
|
|
/*
|
|
* Use slots 3/4 for S/PDIF so SSI won't try to enable
|
|
* other slots and send some samples there
|
|
* due to SLOTREQ bits for S/PDIF received from codec
|
|
*/
|
|
snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
|
|
AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
|
|
codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
|
|
if (ret) {
|
|
dev_err(dev, "failed to set sysclk in %s\n", __func__);
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int fsl_asoc_card_probe(struct platform_device *pdev)
|
|
{
|
|
struct device_node *cpu_np, *codec_np, *asrc_np;
|
|
struct device_node *np = pdev->dev.of_node;
|
|
struct platform_device *asrc_pdev = NULL;
|
|
struct platform_device *cpu_pdev;
|
|
struct fsl_asoc_card_priv *priv;
|
|
struct i2c_client *codec_dev;
|
|
const char *codec_dai_name;
|
|
u32 width;
|
|
int ret;
|
|
|
|
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
|
|
if (!priv)
|
|
return -ENOMEM;
|
|
|
|
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
|
|
/* Give a chance to old DT binding */
|
|
if (!cpu_np)
|
|
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
|
|
if (!cpu_np) {
|
|
dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
|
|
ret = -EINVAL;
|
|
goto fail;
|
|
}
|
|
|
|
cpu_pdev = of_find_device_by_node(cpu_np);
|
|
if (!cpu_pdev) {
|
|
dev_err(&pdev->dev, "failed to find CPU DAI device\n");
|
|
ret = -EINVAL;
|
|
goto fail;
|
|
}
|
|
|
|
codec_np = of_parse_phandle(np, "audio-codec", 0);
|
|
if (codec_np)
|
|
codec_dev = of_find_i2c_device_by_node(codec_np);
|
|
else
|
|
codec_dev = NULL;
|
|
|
|
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
|
|
if (asrc_np)
|
|
asrc_pdev = of_find_device_by_node(asrc_np);
|
|
|
|
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
|
|
if (codec_dev) {
|
|
struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
|
|
|
|
if (!IS_ERR(codec_clk)) {
|
|
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
|
|
clk_put(codec_clk);
|
|
}
|
|
}
|
|
|
|
/* Default sample rate and format, will be updated in hw_params() */
|
|
priv->sample_rate = 44100;
|
|
priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
|
|
|
|
/* Assign a default DAI format, and allow each card to overwrite it */
|
|
priv->dai_fmt = DAI_FMT_BASE;
|
|
|
|
/* Diversify the card configurations */
|
|
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
|
|
codec_dai_name = "cs42888";
|
|
priv->card.set_bias_level = NULL;
|
|
priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
|
|
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
|
|
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
|
|
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
|
|
priv->cpu_priv.slot_width = 32;
|
|
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
|
|
} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
|
|
codec_dai_name = "cs4271-hifi";
|
|
priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
|
|
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
|
|
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
|
|
codec_dai_name = "sgtl5000";
|
|
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
|
|
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
|
|
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
|
|
codec_dai_name = "wm8962";
|
|
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
|
|
priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
|
|
priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
|
|
priv->codec_priv.pll_id = WM8962_FLL;
|
|
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
|
|
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
|
|
codec_dai_name = "wm8960-hifi";
|
|
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
|
|
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
|
|
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
|
|
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
|
|
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
|
|
codec_dai_name = "ac97-hifi";
|
|
priv->card.set_bias_level = NULL;
|
|
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
|
|
} else {
|
|
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
|
|
ret = -EINVAL;
|
|
goto asrc_fail;
|
|
}
|
|
|
|
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
|
|
dev_err(&pdev->dev, "failed to find codec device\n");
|
|
ret = -EINVAL;
|
|
goto asrc_fail;
|
|
}
|
|
|
|
/* Common settings for corresponding Freescale CPU DAI driver */
|
|
if (strstr(cpu_np->name, "ssi")) {
|
|
/* Only SSI needs to configure AUDMUX */
|
|
ret = fsl_asoc_card_audmux_init(np, priv);
|
|
if (ret) {
|
|
dev_err(&pdev->dev, "failed to init audmux\n");
|
|
goto asrc_fail;
|
|
}
|
|
} else if (strstr(cpu_np->name, "esai")) {
|
|
priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
|
|
priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
|
|
} else if (strstr(cpu_np->name, "sai")) {
|
|
priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
|
|
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
|
|
}
|
|
|
|
snprintf(priv->name, sizeof(priv->name), "%s-audio",
|
|
fsl_asoc_card_is_ac97(priv) ? "ac97" :
|
|
codec_dev->name);
|
|
|
|
/* Initialize sound card */
|
|
priv->pdev = pdev;
|
|
priv->card.dev = &pdev->dev;
|
|
priv->card.name = priv->name;
|
|
priv->card.dai_link = priv->dai_link;
|
|
priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
|
|
audio_map_ac97 : audio_map;
|
|
priv->card.late_probe = fsl_asoc_card_late_probe;
|
|
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
|
|
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
|
|
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
|
|
|
|
/* Drop the second half of DAPM routes -- ASRC */
|
|
if (!asrc_pdev)
|
|
priv->card.num_dapm_routes /= 2;
|
|
|
|
memcpy(priv->dai_link, fsl_asoc_card_dai,
|
|
sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
|
|
|
|
ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
|
|
if (ret) {
|
|
dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
|
|
goto asrc_fail;
|
|
}
|
|
|
|
/* Normal DAI Link */
|
|
priv->dai_link[0].cpu_of_node = cpu_np;
|
|
priv->dai_link[0].codec_dai_name = codec_dai_name;
|
|
|
|
if (!fsl_asoc_card_is_ac97(priv))
|
|
priv->dai_link[0].codec_of_node = codec_np;
|
|
else {
|
|
u32 idx;
|
|
|
|
ret = of_property_read_u32(cpu_np, "cell-index", &idx);
|
|
if (ret) {
|
|
dev_err(&pdev->dev,
|
|
"cannot get CPU index property\n");
|
|
goto asrc_fail;
|
|
}
|
|
|
|
priv->dai_link[0].codec_name =
|
|
devm_kasprintf(&pdev->dev, GFP_KERNEL,
|
|
"ac97-codec.%u",
|
|
(unsigned int)idx);
|
|
}
|
|
|
|
priv->dai_link[0].platform_of_node = cpu_np;
|
|
priv->dai_link[0].dai_fmt = priv->dai_fmt;
|
|
priv->card.num_links = 1;
|
|
|
|
if (asrc_pdev) {
|
|
/* DPCM DAI Links only if ASRC exsits */
|
|
priv->dai_link[1].cpu_of_node = asrc_np;
|
|
priv->dai_link[1].platform_of_node = asrc_np;
|
|
priv->dai_link[2].codec_dai_name = codec_dai_name;
|
|
priv->dai_link[2].codec_of_node = codec_np;
|
|
priv->dai_link[2].codec_name =
|
|
priv->dai_link[0].codec_name;
|
|
priv->dai_link[2].cpu_of_node = cpu_np;
|
|
priv->dai_link[2].dai_fmt = priv->dai_fmt;
|
|
priv->card.num_links = 3;
|
|
|
|
ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
|
|
&priv->asrc_rate);
|
|
if (ret) {
|
|
dev_err(&pdev->dev, "failed to get output rate\n");
|
|
ret = -EINVAL;
|
|
goto asrc_fail;
|
|
}
|
|
|
|
ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
|
|
if (ret) {
|
|
dev_err(&pdev->dev, "failed to get output rate\n");
|
|
ret = -EINVAL;
|
|
goto asrc_fail;
|
|
}
|
|
|
|
if (width == 24)
|
|
priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
|
|
else
|
|
priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
|
|
}
|
|
|
|
/* Finish card registering */
|
|
platform_set_drvdata(pdev, priv);
|
|
snd_soc_card_set_drvdata(&priv->card, priv);
|
|
|
|
ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
|
|
if (ret)
|
|
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
|
|
|
|
asrc_fail:
|
|
of_node_put(asrc_np);
|
|
of_node_put(codec_np);
|
|
fail:
|
|
of_node_put(cpu_np);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
|
|
{ .compatible = "fsl,imx-audio-ac97", },
|
|
{ .compatible = "fsl,imx-audio-cs42888", },
|
|
{ .compatible = "fsl,imx-audio-cs427x", },
|
|
{ .compatible = "fsl,imx-audio-sgtl5000", },
|
|
{ .compatible = "fsl,imx-audio-wm8962", },
|
|
{ .compatible = "fsl,imx-audio-wm8960", },
|
|
{}
|
|
};
|
|
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
|
|
|
|
static struct platform_driver fsl_asoc_card_driver = {
|
|
.probe = fsl_asoc_card_probe,
|
|
.driver = {
|
|
.name = "fsl-asoc-card",
|
|
.pm = &snd_soc_pm_ops,
|
|
.of_match_table = fsl_asoc_card_dt_ids,
|
|
},
|
|
};
|
|
module_platform_driver(fsl_asoc_card_driver);
|
|
|
|
MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
|
|
MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
|
|
MODULE_ALIAS("platform:fsl-asoc-card");
|
|
MODULE_LICENSE("GPL");
|