Based on kernel version 6.11
. Page generated on 2024-09-24 08:21 EST
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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 | ======================= ASoC Codec Class Driver ======================= The codec class driver is generic and hardware independent code that configures the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. It should contain no code that is specific to the target platform or machine. All platform and machine specific code should be added to the platform and machine drivers respectively. Each codec class driver *must* provide the following features:- 1. Codec DAI and PCM configuration 2. Codec control IO - using RegMap API 3. Mixers and audio controls 4. Codec audio operations 5. DAPM description. 6. DAPM event handler. Optionally, codec drivers can also provide:- 7. DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec driver code in sound/soc/codecs/ ASoC Codec driver breakdown =========================== Codec DAI and PCM configuration ------------------------------- Each codec driver must have a struct snd_soc_dai_driver to define its DAI and PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. :: static struct snd_soc_dai_ops wm8731_dai_ops = { .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, .mute_stream = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, }; struct snd_soc_dai_driver wm8731_dai = { .name = "wm8731-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, .ops = &wm8731_dai_ops, .symmetric_rate = 1, }; Codec control IO ---------------- The codec can usually be controlled via an I2C or SPI style interface (AC97 combines control with data in the DAI). The codec driver should use the Regmap API for all codec IO. Please see include/linux/regmap.h and existing codec drivers for example regmap usage. Mixers and audio controls ------------------------- All the codec mixers and audio controls can be defined using the convenience macros defined in soc.h. :: #define SOC_SINGLE(xname, reg, shift, mask, invert) Defines a single control as follows:- :: xname = Control name e.g. "Playback Volume" reg = codec register shift = control bit(s) offset in register mask = control bit size(s) e.g. mask of 7 = 3 bits invert = the control is inverted Other macros include:- :: #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) A stereo control :: #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) A stereo control spanning 2 registers :: #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) Defines an single enumerated control as follows:- :: xreg = register xshift = control bit(s) offset in register xmask = control bit(s) size xtexts = pointer to array of strings that describe each setting #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) Defines a stereo enumerated control Codec Audio Operations ---------------------- The codec driver also supports the following ALSA PCM operations:- :: /* SoC audio ops */ struct snd_soc_ops { int (*startup)(struct snd_pcm_substream *); void (*shutdown)(struct snd_pcm_substream *); int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); }; Please refer to the ALSA driver PCM documentation for details. https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html DAPM description ---------------- The Dynamic Audio Power Management description describes the codec power components and their relationships and registers to the ASoC core. Please read dapm.rst for details of building the description. Please also see the examples in other codec drivers. DAPM event handler ------------------ This function is a callback that handles codec domain PM calls and system domain PM calls (e.g. suspend and resume). It is used to put the codec to sleep when not in use. Power states:- :: SNDRV_CTL_POWER_D0: /* full On */ /* vref/mid, clk and osc on, active */ SNDRV_CTL_POWER_D1: /* partial On */ SNDRV_CTL_POWER_D2: /* partial On */ SNDRV_CTL_POWER_D3hot: /* Off, with power */ /* everything off except vref/vmid, inactive */ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ Codec DAC digital mute control ------------------------------ Most codecs have a digital mute before the DACs that can be used to minimise any system noise. The mute stops any digital data from entering the DAC. A callback can be created that is called by the core for each codec DAI when the mute is applied or freed. i.e. :: static int wm8974_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf; if (mute) snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } |