diff options
21 files changed, 784 insertions, 186 deletions
diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt new file mode 100644 index 000000000000..5efc8c068de0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.txt @@ -0,0 +1,13 @@ +GTM601 UMTS modem audio interface CODEC + +This device has no configuration interface. Sample rate is fixed - 8kHz. + +Required properties: + + - compatible : "option,gtm601" + +Example: + +codec: gtm601_codec { + compatible = "option,gtm601"; +}; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index aa802a274520..4e3be6682c98 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -18,6 +18,12 @@ Optional properties: - maxim,dmic-freq: Frequency at which to clock DMIC +- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + Pins on the device (for linking into audio routes): * MIC1 diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 79ad4cbdcdd4..99ffc49aa779 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -23,11 +23,6 @@ #include <sound/soc.h> #include <sound/tlv.h> -struct lm4857 { - struct regmap *regmap; - uint8_t mode; -}; - static const struct reg_default lm4857_default_regs[] = { { 0x0, 0x00 }, { 0x1, 0x00 }, @@ -46,64 +41,33 @@ static const struct reg_default lm4857_default_regs[] = { #define LM4857_WAKEUP 5 #define LM4857_EPGAIN 4 -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - ucontrol->value.integer.value[0] = lm4857->mode; - - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - uint8_t value = ucontrol->value.integer.value[0]; - - lm4857->mode = value; - - if (codec->dapm.bias_level == SND_SOC_BIAS_ON) - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6); - - return 1; -} - -static int lm4857_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - switch (level) { - case SND_SOC_BIAS_ON: - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, - lm4857->mode + 6); - break; - case SND_SOC_BIAS_STANDBY: - regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0); - break; - default: - break; - } - - return 0; -} +static const unsigned int lm4857_mode_values[] = { + 0, + 6, + 7, + 8, + 9, +}; -static const char *lm4857_mode[] = { +static const char * const lm4857_mode_texts[] = { + "Off", "Earpiece", "Loudspeaker", "Loudspeaker + Headphone", "Headphone", }; -static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode); +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(lm4857_mode_enum, + LM4857_CTRL, 0, 0xf, lm4857_mode_texts, lm4857_mode_values); + +static const struct snd_kcontrol_new lm4857_mode_ctrl = + SOC_DAPM_ENUM("Mode", lm4857_mode_enum); static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN"), + SND_SOC_DAPM_DEMUX("Mode", SND_SOC_NOPM, 0, 0, &lm4857_mode_ctrl), + SND_SOC_DAPM_OUTPUT("LS"), SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_OUTPUT("EP"), @@ -125,24 +89,18 @@ static const struct snd_kcontrol_new lm4857_controls[] = { LM4857_WAKEUP, 1, 0), SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL, LM4857_EPGAIN, 1, 0), - - SOC_ENUM_EXT("Mode", lm4857_mode_enum, - lm4857_get_mode, lm4857_set_mode), }; -/* There is a demux between the input signal and the output signals. - * Currently there is no easy way to model it in ASoC and since it does not make - * much of a difference in practice simply connect the input direclty to the - * outputs. */ static const struct snd_soc_dapm_route lm4857_routes[] = { - {"LS", NULL, "IN"}, - {"HP", NULL, "IN"}, - {"EP", NULL, "IN"}, + { "Mode", NULL, "IN" }, + { "LS", "Loudspeaker", "Mode" }, + { "LS", "Loudspeaker + Headphone", "Mode" }, + { "HP", "Headphone", "Mode" }, + { "HP", "Loudspeaker + Headphone", "Mode" }, + { "EP", "Earpiece", "Mode" }, }; -static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { - .set_bias_level = lm4857_set_bias_level, - +static struct snd_soc_component_driver lm4857_component_driver = { .controls = lm4857_controls, .num_controls = ARRAY_SIZE(lm4857_controls), .dapm_widgets = lm4857_dapm_widgets, @@ -165,25 +123,14 @@ static const struct regmap_config lm4857_regmap_config = { static int lm4857_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct lm4857 *lm4857; - - lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); - if (!lm4857) - return -ENOMEM; - - i2c_set_clientdata(i2c, lm4857); - - lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); - if (IS_ERR(lm4857->regmap)) - return PTR_ERR(lm4857->regmap); + struct regmap *regmap; - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); -} + regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); -static int lm4857_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; + return devm_snd_soc_register_component(&i2c->dev, + &lm4857_component_driver, NULL, 0); } static const struct i2c_device_id lm4857_i2c_id[] = { @@ -198,7 +145,6 @@ static struct i2c_driver lm4857_i2c_driver = { .owner = THIS_MODULE, }, .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, .id_table = lm4857_i2c_id, }; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c2306268cab8..679f0a0f7039 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec) struct max98090_cdata *cdata; enum max98090_type devtype; int ret = 0; + int err; + unsigned int micbias; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2503,8 +2505,17 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_MASK); + err = device_property_read_u32(codec->dev, "maxim,micbias", &micbias); + if (err) { + micbias = M98090_MBVSEL_2V8; + dev_info(codec->dev, "use default 2.8v micbias\n"); + } else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) { + dev_err(codec->dev, "micbias out of range 0x%x\n", micbias); + micbias = M98090_MBVSEL_2V8; + } + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, - M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + M98090_MBVSEL_MASK, micbias); max98090_add_widgets(codec); diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index bf3e933ee895..3a2fda08a893 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode"); + sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); return PTR_ERR(sdmode); } - gpiod_direction_output(sdmode, 0); snd_soc_codec_set_drvdata(codec, sdmode); return 0; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ffe6187dce85..60eff36260cb 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ee03dbdda235..791953ffbc41 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -79,7 +79,6 @@ config SND_SOC_INTEL_BROADWELL_MACH depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL - select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell @@ -112,12 +111,24 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" depends on X86_INTEL_LPSS select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5645 audio codec. + platforms with RT5645/5650 audio codec. If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec" + depends on X86_INTEL_LPSS + select SND_SOC_MAX98090 + select SND_SOC_TS3A227E + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with MAX98090 audio codec it also can support TI jack chip as aux device. + If unsure select "N". diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 90aa5c0476f3..61e240935451 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -774,8 +774,120 @@ int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable) return ret; } +int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + ctx->ssp_cmd.nb_slots = slots; + ctx->ssp_cmd.active_tx_slot_map = tx_mask; + ctx->ssp_cmd.active_rx_slot_map = rx_mask; + ctx->ssp_cmd.nb_bits_per_slots = slot_width; + + return 0; +} + +static int sst_get_frame_sync_polarity(struct snd_soc_dai *dai, + unsigned int fmt) +{ + int format; + + format = fmt & SND_SOC_DAIFMT_INV_MASK; + dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); + + switch (format) { + case SND_SOC_DAIFMT_NB_NF: + return SSP_FS_ACTIVE_LOW; + case SND_SOC_DAIFMT_NB_IF: + return SSP_FS_ACTIVE_HIGH; + case SND_SOC_DAIFMT_IB_IF: + return SSP_FS_ACTIVE_LOW; + case SND_SOC_DAIFMT_IB_NF: + return SSP_FS_ACTIVE_HIGH; + default: + dev_err(dai->dev, "Invalid frame sync polarity %d\n", format); + } + + return -EINVAL; +} + +static int sst_get_ssp_mode(struct snd_soc_dai *dai, unsigned int fmt) +{ + int format; + + format = (fmt & SND_SOC_DAIFMT_MASTER_MASK); + dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); + + switch (format) { + case SND_SOC_DAIFMT_CBS_CFS: + return SSP_MODE_MASTER; + case SND_SOC_DAIFMT_CBM_CFM: + return SSP_MODE_SLAVE; + default: + dev_err(dai->dev, "Invalid ssp protocol: %d\n", format); + } + + return -EINVAL; +} + + +int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int mode; + int fs_polarity; + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (mode) { + case SND_SOC_DAIFMT_DSP_B: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1); + ctx->ssp_cmd.start_delay = 0; + ctx->ssp_cmd.data_polarity = 1; + ctx->ssp_cmd.frame_sync_width = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_PCM; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NETWORK << 1); + ctx->ssp_cmd.start_delay = 1; + ctx->ssp_cmd.data_polarity = 1; + ctx->ssp_cmd.frame_sync_width = 1; + break; + + case SND_SOC_DAIFMT_I2S: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1); + ctx->ssp_cmd.start_delay = 1; + ctx->ssp_cmd.data_polarity = 0; + ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots; + break; + + case SND_SOC_DAIFMT_LEFT_J: + ctx->ssp_cmd.ssp_protocol = SSP_MODE_I2S; + ctx->ssp_cmd.mode = sst_get_ssp_mode(dai, fmt) | (SSP_PCM_MODE_NORMAL << 1); + ctx->ssp_cmd.start_delay = 0; + ctx->ssp_cmd.data_polarity = 0; + ctx->ssp_cmd.frame_sync_width = ctx->ssp_cmd.nb_bits_per_slots; + break; + + default: + dev_dbg(dai->dev, "using default ssp configs\n"); + } + + fs_polarity = sst_get_frame_sync_polarity(dai, fmt); + if (fs_polarity < 0) + return fs_polarity; + + ctx->ssp_cmd.frame_sync_polarity = fs_polarity; + + return 0; +} + /** * sst_ssp_config - contains SSP configuration for media UC + * this can be overwritten by set_dai_xxx APIs */ static const struct sst_ssp_config sst_ssp_configs = { .ssp_id = SSP_CODEC, @@ -789,47 +901,56 @@ static const struct sst_ssp_config sst_ssp_configs = { .fs_frequency = SSP_FS_48_KHZ, .active_slot_map = 0xF, .start_delay = 0, + .frame_sync_polarity = SSP_FS_ACTIVE_HIGH, + .data_polarity = 1, }; +void sst_fill_ssp_defaults(struct snd_soc_dai *dai) +{ + const struct sst_ssp_config *config; + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); + + config = &sst_ssp_configs; + + ctx->ssp_cmd.selection = config->ssp_id; + ctx->ssp_cmd.nb_bits_per_slots = config->bits_per_slot; + ctx->ssp_cmd.nb_slots = config->slots; + ctx->ssp_cmd.mode = config->ssp_mode | (config->pcm_mode << 1); + ctx->ssp_cmd.duplex = config->duplex; + ctx->ssp_cmd.active_tx_slot_map = config->active_slot_map; + ctx->ssp_cmd.active_rx_slot_map = config->active_slot_map; + ctx->ssp_cmd.frame_sync_frequency = config->fs_frequency; + ctx->ssp_cmd.frame_sync_polarity = config->frame_sync_polarity; + ctx->ssp_cmd.data_polarity = config->data_polarity; + ctx->ssp_cmd.frame_sync_width = config->fs_width; + ctx->ssp_cmd.ssp_protocol = config->ssp_protocol; + ctx->ssp_cmd.start_delay = config->start_delay; + ctx->ssp_cmd.reserved1 = ctx->ssp_cmd.reserved2 = 0xFF; +} + int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) { - struct sst_cmd_sba_hw_set_ssp cmd; struct sst_data *drv = snd_soc_dai_get_drvdata(dai); const struct sst_ssp_config *config; dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id); - SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); - cmd.header.command_id = SBA_HW_SET_SSP; - cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) + SST_FILL_DEFAULT_DESTINATION(drv->ssp_cmd.header.dst); + drv->ssp_cmd.header.command_id = SBA_HW_SET_SSP; + drv->ssp_cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) - sizeof(struct sst_dsp_header); config = &sst_ssp_configs; dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id); if (enable) - cmd.switch_state = SST_SWITCH_ON; + drv->ssp_cmd.switch_state = SST_SWITCH_ON; else - cmd.switch_state = SST_SWITCH_OFF; - - cmd.selection = config->ssp_id; - cmd.nb_bits_per_slots = config->bits_per_slot; - cmd.nb_slots = config->slots; - cmd.mode = config->ssp_mode | (config->pcm_mode << 1); - cmd.duplex = config->duplex; - cmd.active_tx_slot_map = config->active_slot_map; - cmd.active_rx_slot_map = config->active_slot_map; - cmd.frame_sync_frequency = config->fs_frequency; - cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH; - cmd.data_polarity = 1; - cmd.frame_sync_width = config->fs_width; - cmd.ssp_protocol = config->ssp_protocol; - cmd.start_delay = config->start_delay; - cmd.reserved1 = cmd.reserved2 = 0xFF; + drv->ssp_cmd.switch_state = SST_SWITCH_OFF; return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, - SST_TASK_SBA, 0, &cmd, - sizeof(cmd.header) + cmd.header.length); + SST_TASK_SBA, 0, &drv->ssp_cmd, + sizeof(drv->ssp_cmd.header) + drv->ssp_cmd.header.length); } static int sst_set_be_modules(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index daecc58f28af..93de8045d4e1 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -562,6 +562,8 @@ struct sst_ssp_config { u8 active_slot_map; u8 start_delay; u16 fs_width; + u8 frame_sync_polarity; + u8 data_polarity; }; struct sst_ssp_cfg { @@ -695,7 +697,7 @@ struct sst_gain_mixer_control { u16 module_id; u16 pipe_id; u16 task_id; - char pname[44]; + char pname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; struct snd_soc_dapm_widget *w; }; @@ -867,4 +869,9 @@ struct sst_enum { SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \ SST_SSP_MUX_ENUM(xreg, xshift, xtexts)) +int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width); +int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt); +void sst_fill_ssp_defaults(struct snd_soc_dai *dai); + #endif diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 2fbaf2c75d17..641ebe61dc08 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -434,13 +434,51 @@ static int sst_enable_ssp(struct snd_pcm_substream *substream, if (!dai->active) { ret = sst_handle_vb_timer(dai, true); - if (ret) - return ret; - ret = send_ssp_cmd(dai, dai->name, 1); + sst_fill_ssp_defaults(dai); } return ret; } +static int sst_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + + if (dai->active == 1) + ret = send_ssp_cmd(dai, dai->name, 1); + return ret; +} + +static int sst_set_format(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret = 0; + + if (!dai->active) + return 0; + + ret = sst_fill_ssp_config(dai, fmt); + if (ret < 0) + dev_err(dai->dev, "sst_set_format failed..\n"); + + return ret; +} + +static int sst_platform_set_ssp_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) { + int ret = 0; + + if (!dai->active) + return ret; + + ret = sst_fill_ssp_slot(dai, tx_mask, rx_mask, slots, slot_width); + if (ret < 0) + dev_err(dai->dev, "sst_fill_ssp_slot failed..%d\n", ret); + + return ret; +} + static void sst_disable_ssp(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -465,6 +503,9 @@ static struct snd_soc_dai_ops sst_compr_dai_ops = { static struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, + .hw_params = sst_be_hw_params, + .set_fmt = sst_set_format, + .set_tdm_slot = sst_platform_set_ssp_slot, .shutdown = sst_disable_ssp, }; diff --git a/sound/soc/intel/atom/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h index 9094314be2b0..2409b23eeacf 100644 --- a/sound/soc/intel/atom/sst-mfld-platform.h +++ b/sound/soc/intel/atom/sst-mfld-platform.h @@ -22,6 +22,7 @@ #define __SST_PLATFORMDRV_H__ #include "sst-mfld-dsp.h" +#include "sst-atom-controls.h" extern struct sst_device *sst; @@ -175,6 +176,7 @@ struct sst_data { struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; struct snd_soc_card *soc_card; + struct sst_cmd_sba_hw_set_ssp ssp_cmd; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 05f693083911..bb19b5801466 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -354,6 +354,10 @@ static struct sst_machines sst_acpi_chv[] = { &chv_platform_data }, {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", + &chv_platform_data }, + {"193C9890", "cht-bsw", "cht-bsw-max98090", NULL, + "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index a839dbfa5218..4c01bb43928d 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -679,6 +679,14 @@ static u64 byt_reply_msg_match(u64 header, u64 *mask) return header; } +static bool byt_is_dsp_busy(struct sst_dsp *dsp) +{ + u64 ipcx; + + ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)); +} + int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; @@ -699,6 +707,9 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.shim_dbg = byt_shim_dbg; ipc->ops.tx_data_copy = byt_tx_data_copy; ipc->ops.reply_msg_match = byt_reply_msg_match; + ipc->ops.is_dsp_busy = byt_is_dsp_busy; + ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES; + ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES; err = sst_ipc_init(ipc); if (err != 0) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index f8237f0044eb..cb94895c9edb 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -5,6 +5,7 @@ snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o +snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -13,3 +14,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c new file mode 100644 index 000000000000..1be079423d1e --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -0,0 +1,318 @@ +/* + * cht-bsw-max98090.c - ASoc Machine driver for Intel Cherryview-based + * platforms Cherrytrail and Braswell, with max98090 & TI codec. + * + * Copyright (C) 2015 Intel Corp + * Author: Fang, Yang A <yang.a.fang@intel.com> + * This file is modified from cht_bsw_rt5645.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/acpi.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/max98090.h" +#include "../atom/sst-atom-controls.h" +#include "../../codecs/ts3a227e.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "HiFi" + +struct cht_mc_private { + struct snd_soc_jack jack; + bool ts3a227e_present; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK, + CHT_PLAT_CLK_3_HZ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + int jack_type; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *jack = &ctx->jack; + + /** + * TI supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + if (ctx->ts3a227e_present) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; + + ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + jack_type, jack, NULL, 0); + + if (ret) { + dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + int ret = 0; + unsigned int fmt = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + if (ret < 0) { + dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret); + return ret; + } + + fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret); + return ret; + } + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static int cht_max98090_headset_init(struct snd_soc_component *component) +{ + struct snd_soc_card *card = component->card; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + return ts3a227e_enable_jack_detect(component, &ctx->jack); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_aux_dev cht_max98090_headset_dev = { + .name = "Headset Chip", + .init = cht_max98090_headset_init, + .codec_name = "i2c-104C227E:00", +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtmax98090", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .aux_dev = &cht_max98090_headset_dev, + .num_aux_devs = 1, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + bool found = false; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + if (ACPI_SUCCESS(acpi_get_devices( + "104C227E", + snd_acpi_codec_match, + &found, NULL)) && found) { + drv->ts3a227e_present = true; + } else { + /* no need probe TI jack detection chip */ + snd_soc_card_cht.aux_dev = NULL; + snd_soc_card_cht.num_aux_devs = 0; + drv->ts3a227e_present = false; + } + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-max98090", + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A <yang.a.fang@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-max98090"); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 26e01f36b704..bdcaf467842a 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -21,6 +21,7 @@ */ #include <linux/module.h> +#include <linux/acpi.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/pcm.h> @@ -33,9 +34,15 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5645-aif1" +struct cht_acpi_card { + char *codec_id; + int codec_type; + struct snd_soc_card *soc_card; +}; + struct cht_mc_private { - struct snd_soc_jack hp_jack; - struct snd_soc_jack mic_jack; + struct snd_soc_jack jack; + struct cht_acpi_card *acpi_card; }; static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) @@ -94,7 +101,7 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD), }; -static const struct snd_soc_dapm_route cht_audio_map[] = { +static const struct snd_soc_dapm_route cht_rt5645_audio_map[] = { {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Int Mic"}, @@ -115,6 +122,27 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"Ext Spk", NULL, "Platform Clock"}, }; +static const struct snd_soc_dapm_route cht_rt5650_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L2", NULL, "Int Mic"}, + {"DMIC R2", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -150,6 +178,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; + int jack_type; struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); @@ -169,23 +198,22 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", - SND_JACK_HEADPHONE, &ctx->hp_jack, - NULL, 0); - if (ret) { - dev_err(runtime->dev, "HP jack creation failed %d\n", ret); - return ret; - } + if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", - SND_JACK_MICROPHONE, &ctx->mic_jack, + ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + jack_type, &ctx->jack, NULL, 0); if (ret) { - dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); + dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); + rt5645_set_jack_detect(codec, &ctx->jack, &ctx->jack, &ctx->jack); return ret; } @@ -239,7 +267,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -267,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, + .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, @@ -275,43 +303,85 @@ static struct snd_soc_dai_link cht_dailink[] = { }; /* SoC card */ -static struct snd_soc_card snd_soc_card_cht = { +static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), - .dapm_routes = cht_audio_map, - .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .dapm_routes = cht_rt5645_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_rt5645_audio_map), .controls = cht_mc_controls, .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static struct snd_soc_card snd_soc_card_chtrt5650 = { + .name = "chtrt5650", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_rt5650_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_rt5650_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static struct cht_acpi_card snd_soc_cards[] = { + {"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645}, + {"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650}, +}; + +static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; + int i; struct cht_mc_private *drv; + struct snd_soc_card *card = snd_soc_cards[0].soc_card; + bool found = false; + char codec_name[16]; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) return -ENOMEM; - snd_soc_card_cht.dev = &pdev->dev; - snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { + if (ACPI_SUCCESS(acpi_get_devices( + snd_soc_cards[i].codec_id, + snd_acpi_codec_match, + &found, NULL)) && found) { + dev_dbg(&pdev->dev, + "found codec %s\n", snd_soc_cards[i].codec_id); + card = snd_soc_cards[i].soc_card; + drv->acpi_card = &snd_soc_cards[i]; + break; + } + } + card->dev = &pdev->dev; + sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ + strcpy((char *)card->dai_link[2].codec_name, codec_name); + snd_soc_card_set_drvdata(card, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, &snd_soc_card_cht); + platform_set_drvdata(pdev, card); return ret_val; } static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5645", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 4b62a553823c..a12c7bb08d3b 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -129,11 +129,31 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc) return -ENOMEM; for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + ipc->msg[i].tx_data = kzalloc(ipc->tx_data_max_size, GFP_KERNEL); + if (ipc->msg[i].tx_data == NULL) + goto free_mem; + + ipc->msg[i].rx_data = kzalloc(ipc->rx_data_max_size, GFP_KERNEL); + if (ipc->msg[i].rx_data == NULL) { + kfree(ipc->msg[i].tx_data); + goto free_mem; + } + init_waitqueue_head(&ipc->msg[i].waitq); list_add(&ipc->msg[i].list, &ipc->empty_list); } return 0; + +free_mem: + while (i > 0) { + kfree(ipc->msg[i-1].tx_data); + kfree(ipc->msg[i-1].rx_data); + --i; + } + kfree(ipc->msg); + + return -ENOMEM; } static void ipc_tx_msgs(struct kthread_work *work) @@ -142,7 +162,6 @@ static void ipc_tx_msgs(struct kthread_work *work) container_of(work, struct sst_generic_ipc, kwork); struct ipc_message *msg; unsigned long flags; - u64 ipcx; spin_lock_irqsave(&ipc->dsp->spinlock, flags); @@ -153,8 +172,8 @@ static void ipc_tx_msgs(struct kthread_work *work) /* if the DSP is busy, we will TX messages after IRQ. * also postpone if we are in the middle of procesing completion irq*/ - ipcx = sst_dsp_shim_read_unlocked(ipc->dsp, SST_IPCX); - if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { + if (ipc->ops.is_dsp_busy && ipc->ops.is_dsp_busy(ipc->dsp)) { + dev_dbg(ipc->dev, "ipc_tx_msgs dsp busy\n"); spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); return; } @@ -280,11 +299,18 @@ EXPORT_SYMBOL_GPL(sst_ipc_init); void sst_ipc_fini(struct sst_generic_ipc *ipc) { + int i; + if (ipc->tx_thread) kthread_stop(ipc->tx_thread); - if (ipc->msg) + if (ipc->msg) { + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + kfree(ipc->msg[i].tx_data); + kfree(ipc->msg[i].rx_data); + } kfree(ipc->msg); + } } EXPORT_SYMBOL_GPL(sst_ipc_fini); diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index 125ea451a373..ceb7e468a3fa 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -32,9 +32,9 @@ struct ipc_message { u64 header; /* direction wrt host CPU */ - char tx_data[IPC_MAX_MAILBOX_BYTES]; + char *tx_data; size_t tx_size; - char rx_data[IPC_MAX_MAILBOX_BYTES]; + char *rx_data; size_t rx_size; wait_queue_head_t waitq; @@ -51,6 +51,7 @@ struct sst_plat_ipc_ops { void (*shim_dbg)(struct sst_generic_ipc *, const char *); void (*tx_data_copy)(struct ipc_message *, char *, size_t); u64 (*reply_msg_match)(u64 header, u64 *mask); + bool (*is_dsp_busy)(struct sst_dsp *dsp); }; /* SST generic IPC data */ @@ -68,6 +69,8 @@ struct sst_generic_ipc { struct kthread_work kwork; bool pending; struct ipc_message *msg; + int tx_data_max_size; + int rx_data_max_size; struct sst_plat_ipc_ops ops; }; diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 324eceb07b25..f95f271aab0c 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2098,6 +2098,14 @@ static u64 hsw_reply_msg_match(u64 header, u64 *mask) return header; } +static bool hsw_is_dsp_busy(struct sst_dsp *dsp) +{ + u64 ipcx; + + ipcx = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + return (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)); +} + int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_hsw_ipc_fw_version version; @@ -2117,6 +2125,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) ipc->ops.shim_dbg = hsw_shim_dbg; ipc->ops.tx_data_copy = hsw_tx_data_copy; ipc->ops.reply_msg_match = hsw_reply_msg_match; + ipc->ops.is_dsp_busy = hsw_is_dsp_busy; + + ipc->tx_data_max_size = IPC_MAX_MAILBOX_BYTES; + ipc->rx_data_max_size = IPC_MAX_MAILBOX_BYTES; ret = sst_ipc_init(ipc); if (ret != 0) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae0400d6db..e593e7a4b7a7 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -928,10 +928,15 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - sst_hsw_runtime_module_free(pcm_data->runtime); + if (pcm_data->runtime){ + sst_hsw_runtime_module_free(pcm_data->runtime); + pcm_data->runtime = NULL; + } } - if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { + if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES) && + pdata->runtime_waves) { sst_hsw_runtime_module_free(pdata->runtime_waves); + pdata->runtime_waves = NULL; } } @@ -1204,6 +1209,20 @@ static int hsw_pcm_runtime_idle(struct device *dev) return 0; } +static int hsw_pcm_suspend(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* free all runtime modules */ + hsw_pcm_free_modules(pdata); + /* put the DSP to sleep, fw unloaded after runtime modules freed */ + sst_hsw_dsp_runtime_sleep(hsw); + return 0; +} + static int hsw_pcm_runtime_suspend(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1220,8 +1239,7 @@ static int hsw_pcm_runtime_suspend(struct device *dev) return ret; sst_hsw_set_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES); } - sst_hsw_dsp_runtime_suspend(hsw); - sst_hsw_dsp_runtime_sleep(hsw); + hsw_pcm_suspend(dev); pdata->pm_state = HSW_PM_STATE_RTD3; return 0; @@ -1361,10 +1379,7 @@ static int hsw_pcm_prepare(struct device *dev) if (err < 0) dev_err(dev, "failed to save context for PCM %d\n", i); } - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); + hsw_pcm_suspend(dev); } snd_soc_suspend(pdata->soc_card->dev); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index c2ddf0fbfa28..fded99362d39 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -455,50 +455,36 @@ static int rx51_soc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, pdata); pdata->tvout_selection_gpio = devm_gpiod_get(card->dev, - "tvout-selection"); + "tvout-selection", + GPIOD_OUT_LOW); if (IS_ERR(pdata->tvout_selection_gpio)) { dev_err(card->dev, "could not get tvout selection gpio\n"); return PTR_ERR(pdata->tvout_selection_gpio); } - err = gpiod_direction_output(pdata->tvout_selection_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup tvout selection gpio\n"); - return err; - } - pdata->jack_detection_gpio = devm_gpiod_get(card->dev, - "jack-detection"); + "jack-detection", + GPIOD_ASIS); if (IS_ERR(pdata->jack_detection_gpio)) { dev_err(card->dev, "could not get jack detection gpio\n"); return PTR_ERR(pdata->jack_detection_gpio); } - pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); + pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch", + GPIOD_OUT_HIGH); if (IS_ERR(pdata->eci_sw_gpio)) { dev_err(card->dev, "could not get eci switch gpio\n"); return PTR_ERR(pdata->eci_sw_gpio); } - err = gpiod_direction_output(pdata->eci_sw_gpio, 1); - if (err) { - dev_err(card->dev, "could not setup eci switch gpio\n"); - return err; - } - pdata->speaker_amp_gpio = devm_gpiod_get(card->dev, - "speaker-amplifier"); + "speaker-amplifier", + GPIOD_OUT_LOW); if (IS_ERR(pdata->speaker_amp_gpio)) { dev_err(card->dev, "could not get speaker enable gpio\n"); return PTR_ERR(pdata->speaker_amp_gpio); } - err = gpiod_direction_output(pdata->speaker_amp_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup speaker enable gpio\n"); - return err; - } - err = devm_snd_soc_register_card(card->dev, card); if (err) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); |