| Author | Tokens | Token Proportion | Commits | Commit Proportion |
|---|---|---|---|---|
| Kuninori Morimoto | 3268 | 97.96% | 22 | 68.75% |
| Ryo Kodama | 28 | 0.84% | 1 | 3.12% |
| Geert Uytterhoeven | 26 | 0.78% | 2 | 6.25% |
| Gustavo A. R. Silva | 6 | 0.18% | 1 | 3.12% |
| Stephen Kitt | 2 | 0.06% | 1 | 3.12% |
| Axel Lin | 2 | 0.06% | 1 | 3.12% |
| Rob Herring | 1 | 0.03% | 1 | 3.12% |
| Biju Das | 1 | 0.03% | 1 | 3.12% |
| Bhumika Goyal | 1 | 0.03% | 1 | 3.12% |
| Uwe Kleine-König | 1 | 0.03% | 1 | 3.12% |
| Total | 3336 | 32 |
// SPDX-License-Identifier: GPL-2.0 // // ak4613.c -- Asahi Kasei ALSA Soc Audio driver // // Copyright (C) 2015 Renesas Electronics Corporation // Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> // // Based on ak4642.c by Kuninori Morimoto // Based on wm8731.c by Richard Purdie // Based on ak4535.c by Richard Purdie // Based on wm8753.c by Liam Girdwood /* * +-------+ * |AK4613 | * SDTO1 <-| | * | | * SDTI1 ->| | * SDTI2 ->| | * SDTI3 ->| | * +-------+ * * +---+ * clk | |___________________________________________... * * [TDM512] * SDTO1 [L1][R1][L2][R2] * SDTI1 [L1][R1][L2][R2][L3][R3][L4][R4][L5][R5][L6][R6] * * [TDM256] * SDTO1 [L1][R1][L2][R2] * SDTI1 [L1][R1][L2][R2][L3][R3][L4][R4] * SDTI2 [L5][R5][L6][R6] * * [TDM128] * SDTO1 [L1][R1][L2][R2] * SDTI1 [L1][R1][L2][R2] * SDTI2 [L3][R3][L4][R4] * SDTI3 [L5][R5][L6][R6] * * [STEREO] * Playback 2ch : SDTI1 * Capture 2ch : SDTO1 * * [TDM512] * Playback 12ch : SDTI1 * Capture 4ch : SDTO1 * * [TDM256] * Playback 12ch : SDTI1 + SDTI2 * Playback 8ch : SDTI1 * Capture 4ch : SDTO1 * * [TDM128] * Playback 12ch : SDTI1 + SDTI2 + SDTI3 * Playback 8ch : SDTI1 + SDTI2 * Playback 4ch : SDTI1 * Capture 4ch : SDTO1 * * * !!! NOTE !!! * * Renesas is the only user of ak4613 on upstream so far, * but the chip connection is like below. * Thus, Renesas can't test all connection case. * Tested TDM is very limited. * * +-----+ +-----------+ * | SoC | | AK4613 | * | |<-----|SDTO1 IN1|<-- Mic * | | | IN2| * | | | | * | |----->|SDTI1 OUT1|--> Headphone * +-----+ |SDTI2 OUT2| * |SDTI3 OUT3| * | OUT4| * | OUT5| * | OUT6| * +-----------+ * * Renesas SoC can handle [2, 6,8] channels. * Ak4613 can handle [2,4, 8,12] channels. * * Because of above HW connection and available channels number, * Renesas could test are ... * * [STEREO] Playback 2ch : SDTI1 * Capture 2ch : SDTO1 * [TDM256] Playback 8ch : SDTI1 (*) * * (*) it used 8ch data between SoC <-> AK4613 on TDM256 mode, * but could confirm is only first 2ch because only 1 * Headphone is connected. * * see * AK4613_ENABLE_TDM_TEST */ #include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of.h> #include <linux/of_graph.h> #include <linux/module.h> #include <linux/regmap.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <sound/tlv.h> #define PW_MGMT1 0x00 /* Power Management 1 */ #define PW_MGMT2 0x01 /* Power Management 2 */ #define PW_MGMT3 0x02 /* Power Management 3 */ #define CTRL1 0x03 /* Control 1 */ #define CTRL2 0x04 /* Control 2 */ #define DEMP1 0x05 /* De-emphasis1 */ #define DEMP2 0x06 /* De-emphasis2 */ #define OFD 0x07 /* Overflow Detect */ #define ZRD 0x08 /* Zero Detect */ #define ICTRL 0x09 /* Input Control */ #define OCTRL 0x0a /* Output Control */ #define LOUT1 0x0b /* LOUT1 Volume Control */ #define ROUT1 0x0c /* ROUT1 Volume Control */ #define LOUT2 0x0d /* LOUT2 Volume Control */ #define ROUT2 0x0e /* ROUT2 Volume Control */ #define LOUT3 0x0f /* LOUT3 Volume Control */ #define ROUT3 0x10 /* ROUT3 Volume Control */ #define LOUT4 0x11 /* LOUT4 Volume Control */ #define ROUT4 0x12 /* ROUT4 Volume Control */ #define LOUT5 0x13 /* LOUT5 Volume Control */ #define ROUT5 0x14 /* ROUT5 Volume Control */ #define LOUT6 0x15 /* LOUT6 Volume Control */ #define ROUT6 0x16 /* ROUT6 Volume Control */ /* PW_MGMT1 */ #define RSTN BIT(0) #define PMDAC BIT(1) #define PMADC BIT(2) #define PMVR BIT(3) /* PW_MGMT2 */ #define PMAD_ALL 0x7 /* PW_MGMT3 */ #define PMDA_ALL 0x3f /* CTRL1 */ #define DIF0 BIT(3) #define DIF1 BIT(4) #define DIF2 BIT(5) #define TDM0 BIT(6) #define TDM1 BIT(7) #define NO_FMT (0xff) #define FMT_MASK (0xf8) /* CTRL2 */ #define DFS_MASK (3 << 2) #define DFS_NORMAL_SPEED (0 << 2) #define DFS_DOUBLE_SPEED (1 << 2) #define DFS_QUAD_SPEED (2 << 2) /* ICTRL */ #define ICTRL_MASK (0x3) /* OCTRL */ #define OCTRL_MASK (0x3F) /* * configs * * 0x000000BA * * B : AK4613_CONFIG_SDTI_x * A : AK4613_CONFIG_MODE_x */ #define AK4613_CONFIG_SET(priv, x) priv->configs |= AK4613_CONFIG_##x #define AK4613_CONFIG_GET(priv, x) (priv->configs & AK4613_CONFIG_##x##_MASK) /* * AK4613_CONFIG_SDTI_x * * It indicates how many SDTIx is connected. */ #define AK4613_CONFIG_SDTI_MASK (0xF << 4) #define AK4613_CONFIG_SDTI(x) (((x) & 0xF) << 4) #define AK4613_CONFIG_SDTI_set(priv, x) AK4613_CONFIG_SET(priv, SDTI(x)) #define AK4613_CONFIG_SDTI_get(priv) ((AK4613_CONFIG_GET(priv, SDTI) >> 4) & 0xF) /* * AK4613_CONFIG_MODE_x * * Same as Ctrl1 :: TDM1/TDM0 * No shift is requested * see * AK4613_CTRL1_TO_MODE() * Table 11/12/13/14 */ #define AK4613_CONFIG_MODE_MASK (0xF) #define AK4613_CONFIG_MODE_STEREO (0x0) #define AK4613_CONFIG_MODE_TDM512 (0x1) #define AK4613_CONFIG_MODE_TDM256 (0x2) #define AK4613_CONFIG_MODE_TDM128 (0x3) /* * !!!! FIXME !!!! * * Because of testable HW limitation, TDM256 8ch TDM was only tested. * This driver uses AK4613_ENABLE_TDM_TEST instead of new DT property so far. * Don't hesitate to update driver, you don't need to care compatible * with Renesas. * * #define AK4613_ENABLE_TDM_TEST */ struct ak4613_interface { unsigned int width; unsigned int fmt; u8 dif; }; struct ak4613_priv { struct mutex lock; struct snd_pcm_hw_constraint_list constraint_rates; struct snd_pcm_hw_constraint_list constraint_channels; struct work_struct dummy_write_work; struct snd_soc_component *component; unsigned int rate; unsigned int sysclk; unsigned int fmt; unsigned int configs; int cnt; u8 ctrl1; u8 oc; u8 ic; }; /* * Playback Volume * * max : 0x00 : 0 dB * ( 0.5 dB step ) * min : 0xFE : -127.0 dB * mute: 0xFF */ static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1); static const struct snd_kcontrol_new ak4613_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1, 0, 0xFF, 1, out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2, 0, 0xFF, 1, out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3, 0, 0xFF, 1, out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4, 0, 0xFF, 1, out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5, 0, 0xFF, 1, out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6, 0, 0xFF, 1, out_tlv), }; static const struct reg_default ak4613_reg[] = { { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, }; /* * CTRL1 register * see * Table 11/12/13/14 */ #define AUDIO_IFACE(_dif, _width, _fmt) \ { \ .dif = _dif, \ .width = _width, \ .fmt = SND_SOC_DAIFMT_##_fmt,\ } static const struct ak4613_interface ak4613_iface[] = { /* It doesn't support asymmetric format */ AUDIO_IFACE(0x03, 24, LEFT_J), AUDIO_IFACE(0x04, 24, I2S), }; #define AK4613_CTRL1_TO_MODE(priv) ((priv)->ctrl1 >> 6) /* AK4613_CONFIG_MODE_x */ static const struct regmap_config ak4613_regmap_cfg = { .reg_bits = 8, .val_bits = 8, .max_register = 0x16, .reg_defaults = ak4613_reg, .num_reg_defaults = ARRAY_SIZE(ak4613_reg), .cache_type = REGCACHE_RBTREE, }; static const struct of_device_id ak4613_of_match[] = { { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, {}, }; MODULE_DEVICE_TABLE(of, ak4613_of_match); static const struct i2c_device_id ak4613_i2c_id[] = { { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, { } }; MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { /* Outputs */ SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("LOUT3"), SND_SOC_DAPM_OUTPUT("LOUT4"), SND_SOC_DAPM_OUTPUT("LOUT5"), SND_SOC_DAPM_OUTPUT("LOUT6"), SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("ROUT2"), SND_SOC_DAPM_OUTPUT("ROUT3"), SND_SOC_DAPM_OUTPUT("ROUT4"), SND_SOC_DAPM_OUTPUT("ROUT5"), SND_SOC_DAPM_OUTPUT("ROUT6"), /* Inputs */ SND_SOC_DAPM_INPUT("LIN1"), SND_SOC_DAPM_INPUT("LIN2"), SND_SOC_DAPM_INPUT("RIN1"), SND_SOC_DAPM_INPUT("RIN2"), /* DAC */ SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), /* ADC */ SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), }; static const struct snd_soc_dapm_route ak4613_intercon[] = { {"LOUT1", NULL, "DAC1"}, {"LOUT2", NULL, "DAC2"}, {"LOUT3", NULL, "DAC3"}, {"LOUT4", NULL, "DAC4"}, {"LOUT5", NULL, "DAC5"}, {"LOUT6", NULL, "DAC6"}, {"ROUT1", NULL, "DAC1"}, {"ROUT2", NULL, "DAC2"}, {"ROUT3", NULL, "DAC3"}, {"ROUT4", NULL, "DAC4"}, {"ROUT5", NULL, "DAC5"}, {"ROUT6", NULL, "DAC6"}, {"DAC1", NULL, "Playback"}, {"DAC2", NULL, "Playback"}, {"DAC3", NULL, "Playback"}, {"DAC4", NULL, "Playback"}, {"DAC5", NULL, "Playback"}, {"DAC6", NULL, "Playback"}, {"Capture", NULL, "ADC1"}, {"Capture", NULL, "ADC2"}, {"ADC1", NULL, "LIN1"}, {"ADC2", NULL, "LIN2"}, {"ADC1", NULL, "RIN1"}, {"ADC2", NULL, "RIN2"}, }; static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); struct device *dev = component->dev; mutex_lock(&priv->lock); priv->cnt--; if (priv->cnt < 0) { dev_err(dev, "unexpected counter error\n"); priv->cnt = 0; } if (!priv->cnt) priv->ctrl1 = 0; mutex_unlock(&priv->lock); } static void ak4613_hw_constraints(struct ak4613_priv *priv, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; static const unsigned int ak4613_rates[] = { 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, }; #define AK4613_CHANNEL_2 0 #define AK4613_CHANNEL_4 1 #define AK4613_CHANNEL_8 2 #define AK4613_CHANNEL_12 3 #define AK4613_CHANNEL_NONE -1 static const unsigned int ak4613_channels[] = { [AK4613_CHANNEL_2] = 2, [AK4613_CHANNEL_4] = 4, [AK4613_CHANNEL_8] = 8, [AK4613_CHANNEL_12] = 12, }; #define MODE_MAX 4 #define SDTx_MAX 4 #define MASK(x) (1 << AK4613_CHANNEL_##x) static const int mask_list[MODE_MAX][SDTx_MAX] = { /* SDTO SDTIx1 SDTIx2 SDTIx3 */ [AK4613_CONFIG_MODE_STEREO] = { MASK(2), MASK(2), MASK(2), MASK(2)}, [AK4613_CONFIG_MODE_TDM512] = { MASK(4), MASK(12), MASK(12), MASK(12)}, [AK4613_CONFIG_MODE_TDM256] = { MASK(4), MASK(8), MASK(8)|MASK(12), MASK(8)|MASK(12)}, [AK4613_CONFIG_MODE_TDM128] = { MASK(4), MASK(4), MASK(4)|MASK(8), MASK(4)|MASK(8)|MASK(12)}, }; struct snd_pcm_hw_constraint_list *constraint; unsigned int mask; unsigned int mode; unsigned int fs; int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int sdti_num; int i; constraint = &priv->constraint_rates; constraint->list = ak4613_rates; constraint->mask = 0; constraint->count = 0; /* * Slave Mode * Normal: [32kHz, 48kHz] : 256fs,384fs or 512fs * Double: [64kHz, 96kHz] : 256fs * Quad : [128kHz,192kHz]: 128fs * * Master mode * Normal: [32kHz, 48kHz] : 256fs or 512fs * Double: [64kHz, 96kHz] : 256fs * Quad : [128kHz,192kHz]: 128fs */ for (i = 0; i < ARRAY_SIZE(ak4613_rates); i++) { /* minimum fs on each range */ fs = (ak4613_rates[i] <= 96000) ? 256 : 128; if (priv->sysclk >= ak4613_rates[i] * fs) constraint->count = i + 1; } snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, constraint); sdti_num = AK4613_CONFIG_SDTI_get(priv); if (WARN_ON(sdti_num >= SDTx_MAX)) return; if (priv->cnt) { /* * If it was already working, * the constraint is same as working mode. */ mode = AK4613_CTRL1_TO_MODE(priv); mask = 0; /* no default */ } else { /* * It is not yet working, * the constraint is based on board configs. * STEREO mask is default */ mode = AK4613_CONFIG_GET(priv, MODE); mask = mask_list[AK4613_CONFIG_MODE_STEREO][is_play * sdti_num]; } if (WARN_ON(mode >= MODE_MAX)) return; /* add each mode mask */ mask |= mask_list[mode][is_play * sdti_num]; constraint = &priv->constraint_channels; constraint->list = ak4613_channels; constraint->mask = mask; constraint->count = sizeof(ak4613_channels); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, constraint); } static int ak4613_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); mutex_lock(&priv->lock); ak4613_hw_constraints(priv, substream); priv->cnt++; mutex_unlock(&priv->lock); return 0; } static int ak4613_dai_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_component *component = codec_dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); priv->sysclk = freq; return 0; } static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); unsigned int fmt; fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; break; default: return -EINVAL; } fmt = format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; switch (fmt) { case SND_SOC_DAIFMT_CBC_CFC: break; default: /* * SUPPORTME * * "clock provider" is not yet supperted */ return -EINVAL; } return 0; } static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); struct device *dev = component->dev; unsigned int width = params_width(params); unsigned int fmt = priv->fmt; unsigned int rate; int i, ret; u8 ctrl2; rate = params_rate(params); switch (rate) { case 32000: case 44100: case 48000: ctrl2 = DFS_NORMAL_SPEED; break; case 64000: case 88200: case 96000: ctrl2 = DFS_DOUBLE_SPEED; break; case 176400: case 192000: ctrl2 = DFS_QUAD_SPEED; break; default: return -EINVAL; } priv->rate = rate; /* * FIXME * * It doesn't have full TDM suppert yet */ ret = -EINVAL; mutex_lock(&priv->lock); if (priv->cnt > 1) { /* * If it was already working, use current priv->ctrl1 */ ret = 0; } else { /* * It is not yet working, */ unsigned int channel = params_channels(params); u8 tdm; /* STEREO or TDM */ if (channel == 2) tdm = AK4613_CONFIG_MODE_STEREO; else tdm = AK4613_CONFIG_GET(priv, MODE); for (i = ARRAY_SIZE(ak4613_iface) - 1; i >= 0; i--) { const struct ak4613_interface *iface = ak4613_iface + i; if ((iface->fmt == fmt) && (iface->width == width)) { /* * Ctrl1 * | D7 | D6 | D5 | D4 | D3 | D2 | D1 | D0 | * |TDM1|TDM0|DIF2|DIF1|DIF0|ATS1|ATS0|SMUTE| * < tdm > < iface->dif > */ priv->ctrl1 = (tdm << 6) | (iface->dif << 3); ret = 0; break; } } } mutex_unlock(&priv->lock); if (ret < 0) goto hw_params_end; snd_soc_component_update_bits(component, CTRL1, FMT_MASK, priv->ctrl1); snd_soc_component_update_bits(component, CTRL2, DFS_MASK, ctrl2); snd_soc_component_update_bits(component, ICTRL, ICTRL_MASK, priv->ic); snd_soc_component_update_bits(component, OCTRL, OCTRL_MASK, priv->oc); hw_params_end: if (ret < 0) dev_warn(dev, "unsupported data width/format combination\n"); return ret; } static int ak4613_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { u8 mgmt1 = 0; switch (level) { case SND_SOC_BIAS_ON: mgmt1 |= RSTN; fallthrough; case SND_SOC_BIAS_PREPARE: mgmt1 |= PMADC | PMDAC; fallthrough; case SND_SOC_BIAS_STANDBY: mgmt1 |= PMVR; fallthrough; case SND_SOC_BIAS_OFF: default: break; } snd_soc_component_write(component, PW_MGMT1, mgmt1); return 0; } static void ak4613_dummy_write(struct work_struct *work) { struct ak4613_priv *priv = container_of(work, struct ak4613_priv, dummy_write_work); struct snd_soc_component *component = priv->component; unsigned int mgmt1; unsigned int mgmt3; /* * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks * * Note * * To avoid extra delay, we want to avoid preemption here, * but we can't. Because it uses I2C access which is using IRQ * and sleep. Thus, delay might be more than 5 LR clocks * see also * ak4613_dai_trigger() */ udelay(5000000 / priv->rate); mgmt1 = snd_soc_component_read(component, PW_MGMT1); mgmt3 = snd_soc_component_read(component, PW_MGMT3); snd_soc_component_write(component, PW_MGMT1, mgmt1); snd_soc_component_write(component, PW_MGMT3, mgmt3); } static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; struct ak4613_priv *priv = snd_soc_component_get_drvdata(component); /* * FIXME * * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks * from Power Down Release. Otherwise, Playback volume will be 0dB. * To avoid complex multiple delay/dummy_write method from * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), * call it once here. * * But, unfortunately, we can't "write" here because here is atomic * context (It uses I2C access for writing). * Thus, use schedule_work() to switching to normal context * immediately. * * Note * * Calling ak4613_dummy_write() function might be delayed. * In such case, ak4613 volume might be temporarily 0dB when * beggining of playback. * see also * ak4613_dummy_write() */ if ((cmd != SNDRV_PCM_TRIGGER_START) && (cmd != SNDRV_PCM_TRIGGER_RESUME)) return 0; if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return 0; priv->component = component; schedule_work(&priv->dummy_write_work); return 0; } /* * Select below from Sound Card, not Auto * SND_SOC_DAIFMT_CBC_CFC * SND_SOC_DAIFMT_CBP_CFP */ static u64 ak4613_dai_formats = SND_SOC_POSSIBLE_DAIFMT_I2S | SND_SOC_POSSIBLE_DAIFMT_LEFT_J; static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, .auto_selectable_formats = &ak4613_dai_formats, .num_auto_selectable_formats = 1, }; #define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_64000 |\ SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { .name = "ak4613-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 12, .rates = AK4613_PCM_RATE, .formats = AK4613_PCM_FMTBIT, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 4, .rates = AK4613_PCM_RATE, .formats = AK4613_PCM_FMTBIT, }, .ops = &ak4613_dai_ops, .symmetric_rate = 1, }; static int ak4613_suspend(struct snd_soc_component *component) { struct regmap *regmap = dev_get_regmap(component->dev, NULL); regcache_cache_only(regmap, true); regcache_mark_dirty(regmap); return 0; } static int ak4613_resume(struct snd_soc_component *component) { struct regmap *regmap = dev_get_regmap(component->dev, NULL); regcache_cache_only(regmap, false); return regcache_sync(regmap); } static const struct snd_soc_component_driver soc_component_dev_ak4613 = { .suspend = ak4613_suspend, .resume = ak4613_resume, .set_bias_level = ak4613_set_bias_level, .controls = ak4613_snd_controls, .num_controls = ARRAY_SIZE(ak4613_snd_controls), .dapm_widgets = ak4613_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), .dapm_routes = ak4613_intercon, .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), .idle_bias_on = 1, .endianness = 1, }; static void ak4613_parse_of(struct ak4613_priv *priv, struct device *dev) { struct device_node *np = dev->of_node; char prop[32]; int sdti_num; int i; /* Input 1 - 2 */ for (i = 0; i < 2; i++) { snprintf(prop, sizeof(prop), "asahi-kasei,in%d-single-end", i + 1); if (!of_get_property(np, prop, NULL)) priv->ic |= 1 << i; } /* Output 1 - 6 */ for (i = 0; i < 6; i++) { snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1); if (!of_get_property(np, prop, NULL)) priv->oc |= 1 << i; } /* * enable TDM256 test * * !!! FIXME !!! * * It should be configured by DT or other way * if it was full supported. * But it is using ifdef style for now for test * purpose. */ #if defined(AK4613_ENABLE_TDM_TEST) AK4613_CONFIG_SET(priv, MODE_TDM256); #endif /* * connected STDI * TDM support is assuming it is probed via Audio-Graph-Card style here. * Default is SDTIx1 if it was probed via Simple-Audio-Card for now. */ sdti_num = of_graph_get_endpoint_count(np); if ((sdti_num >= SDTx_MAX) || (sdti_num < 1)) sdti_num = 1; AK4613_CONFIG_SDTI_set(priv, sdti_num); } static int ak4613_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; const struct regmap_config *regmap_cfg; struct regmap *regmap; struct ak4613_priv *priv; regmap_cfg = i2c_get_match_data(i2c); if (!regmap_cfg) return -EINVAL; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; ak4613_parse_of(priv, dev); priv->ctrl1 = 0; priv->cnt = 0; priv->sysclk = 0; INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); i2c_set_clientdata(i2c, priv); regmap = devm_regmap_init_i2c(i2c, regmap_cfg); if (IS_ERR(regmap)) return PTR_ERR(regmap); return devm_snd_soc_register_component(dev, &soc_component_dev_ak4613, &ak4613_dai, 1); } static struct i2c_driver ak4613_i2c_driver = { .driver = { .name = "ak4613-codec", .of_match_table = ak4613_of_match, }, .probe = ak4613_i2c_probe, .id_table = ak4613_i2c_id, }; module_i2c_driver(ak4613_i2c_driver); MODULE_DESCRIPTION("Soc AK4613 driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); MODULE_LICENSE("GPL v2");
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