From 6610550c4c2663f51cec308a88870da20db48113 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 3 Nov 2015 15:08:35 +0000 Subject: [PATCH 01/23] ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs This patch adds support for the Cirrus Logic CS47L24 and WM1831 codecs. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l24.c | 1148 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 4 files changed, 1181 insertions(+) create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4c11ea..55e14a3ed5e1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -195,10 +196,12 @@ config SND_SOC_88PM860X config SND_SOC_ARIZONA tristate + default y if SND_SOC_CS47L24=y default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y default y if SND_SOC_WM8997=y default y if SND_SOC_WM8998=y + default m if SND_SOC_CS47L24=m default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m default m if SND_SOC_WM8997=m @@ -211,9 +214,11 @@ config SND_SOC_WM_HUBS config SND_SOC_WM_ADSP tristate + default y if SND_SOC_CS47L24=y default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y default y if SND_SOC_WM2200=y + default m if SND_SOC_CS47L24=m default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m default m if SND_SOC_WM2200=m @@ -422,6 +427,9 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS47L24 + tristate + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f632fc42f59f..c1d73fe4cf6b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs47l24-objs := cs47l24.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -242,6 +243,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c new file mode 100644 index 000000000000..dc5ae7f7a1bd --- /dev/null +++ b/sound/soc/codecs/cs47l24.c @@ -0,0 +1,1148 @@ +/* + * cs47l24.h -- ALSA SoC Audio driver for Cirrus Logic CS47L24 + * + * Copyright 2015 Cirrus Logic Inc. + * + * Author: Richard Fitzgerald + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" +#include "wm_adsp.h" +#include "cs47l24.h" + +struct cs47l24_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static const struct wm_adsp_region cs47l24_dsp2_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x200000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x280000 }, + { .type = WMFW_ADSP2_XM, .base = 0x290000 }, + { .type = WMFW_ADSP2_YM, .base = 0x2a8000 }, +}; + +static const struct wm_adsp_region cs47l24_dsp3_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x300000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x380000 }, + { .type = WMFW_ADSP2_XM, .base = 0x390000 }, + { .type = WMFW_ADSP2_YM, .base = 0x3a8000 }, +}; + +static const struct wm_adsp_region *cs47l24_dsp_regions[] = { + cs47l24_dsp2_regions, + cs47l24_dsp3_regions, +}; + +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +#define CS47L24_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0) + +static const struct snd_kcontrol_new cs47l24_snd_controls[] = { +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), + +SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2L HPF Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2R HPF Switch", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_HPF_SHIFT, 1, 0), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), + +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), +SOC_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), +SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1), + +ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 SC Protect Switch", ARIZONA_HP1_SHORT_CIRCUIT_CTRL, + ARIZONA_HP1_SC_ENA_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +CS47L24_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +CS47L24_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +CS47L24_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP2L, ARIZONA_DSP2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP2R, ARIZONA_DSP2RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP2, ARIZONA_DSP2AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DSP3L, ARIZONA_DSP3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DSP3R, ARIZONA_DSP3RMIX_INPUT_1_SOURCE); +ARIZONA_DSP_AUX_ENUMS(DSP3, ARIZONA_DSP3AUX1MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT3, ARIZONA_ISRC2INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT4, ARIZONA_ISRC2INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC3, ARIZONA_ISRC2DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC4, ARIZONA_ISRC2DEC4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3INT1, ARIZONA_ISRC3INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT2, ARIZONA_ISRC3INT2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT3, ARIZONA_ISRC3INT3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3INT4, ARIZONA_ISRC3INT4MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC3DEC1, ARIZONA_ISRC3DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE); + +static const char * const cs47l24_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "SPKOUT", +}; + +static const unsigned int cs47l24_aec_loopback_values[] = { + 0, 1, 6, +}; + +static const struct soc_enum cs47l24_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(cs47l24_aec_loopback_texts), + cs47l24_aec_loopback_texts, + cs47l24_aec_loopback_values); + +static const struct snd_kcontrol_new cs47l24_aec_loopback_mux = + SOC_DAPM_ENUM("AEC Loopback", cs47l24_aec_loopback); + +static const struct snd_soc_dapm_widget cs47l24_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, + ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +WM_ADSP2("DSP2", 1), +WM_ADSP2("DSP3", 2), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC3", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC4", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3INT1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3INT4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_INT3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC3DEC1", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC2", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC3", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, + ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &cs47l24_aec_loopback_mux), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), +ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"), +ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"), + +ARIZONA_DSP_WIDGETS(DSP2, "DSP2"), +ARIZONA_DSP_WIDGETS(DSP3, "DSP3"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), +ARIZONA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"), +ARIZONA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), +ARIZONA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"), +ARIZONA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"), + +ARIZONA_MUX_WIDGETS(ISRC3DEC1, "ISRC3DEC1"), +ARIZONA_MUX_WIDGETS(ISRC3DEC2, "ISRC3DEC2"), +ARIZONA_MUX_WIDGETS(ISRC3DEC3, "ISRC3DEC3"), +ARIZONA_MUX_WIDGETS(ISRC3DEC4, "ISRC3DEC4"), + +ARIZONA_MUX_WIDGETS(ISRC3INT1, "ISRC3INT1"), +ARIZONA_MUX_WIDGETS(ISRC3INT2, "ISRC3INT2"), +ARIZONA_MUX_WIDGETS(ISRC3INT3, "ISRC3INT3"), +ARIZONA_MUX_WIDGETS(ISRC3INT4, "ISRC3INT4"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF2RX5", "AIF2RX5" }, \ + { name, "AIF2RX6", "AIF2RX6" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2DEC3", "ISRC2DEC3" }, \ + { name, "ISRC2DEC4", "ISRC2DEC4" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ + { name, "ISRC2INT3", "ISRC2INT3" }, \ + { name, "ISRC2INT4", "ISRC2INT4" }, \ + { name, "ISRC3DEC1", "ISRC3DEC1" }, \ + { name, "ISRC3DEC2", "ISRC3DEC2" }, \ + { name, "ISRC3DEC3", "ISRC3DEC3" }, \ + { name, "ISRC3DEC4", "ISRC3DEC4" }, \ + { name, "ISRC3INT1", "ISRC3INT1" }, \ + { name, "ISRC3INT2", "ISRC3INT2" }, \ + { name, "ISRC3INT3", "ISRC3INT3" }, \ + { name, "ISRC3INT4", "ISRC3INT4" }, \ + { name, "DSP2.1", "DSP2" }, \ + { name, "DSP2.2", "DSP2" }, \ + { name, "DSP2.3", "DSP2" }, \ + { name, "DSP2.4", "DSP2" }, \ + { name, "DSP2.5", "DSP2" }, \ + { name, "DSP2.6", "DSP2" }, \ + { name, "DSP3.1", "DSP3" }, \ + { name, "DSP3.2", "DSP3" }, \ + { name, "DSP3.3", "DSP3" }, \ + { name, "DSP3.4", "DSP3" }, \ + { name, "DSP3.5", "DSP3" }, \ + { name, "DSP3.6", "DSP3" } + +static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + { "AIF2 Capture", NULL, "AIF2TX5" }, + { "AIF2 Capture", NULL, "AIF2TX6" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + { "AIF2RX5", NULL, "AIF2 Playback" }, + { "AIF2RX6", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"), + ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + + ARIZONA_DSP_ROUTES("DSP2"), + ARIZONA_DSP_ROUTES("DSP3"), + + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"), + ARIZONA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"), + ARIZONA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"), + + ARIZONA_MUX_ROUTES("ISRC3INT1", "ISRC3INT1"), + ARIZONA_MUX_ROUTES("ISRC3INT2", "ISRC3INT2"), + ARIZONA_MUX_ROUTES("ISRC3INT3", "ISRC3INT3"), + ARIZONA_MUX_ROUTES("ISRC3INT4", "ISRC3INT4"), + + ARIZONA_MUX_ROUTES("ISRC3DEC1", "ISRC3DEC1"), + ARIZONA_MUX_ROUTES("ISRC3DEC2", "ISRC3DEC2"), + ARIZONA_MUX_ROUTES("ISRC3DEC3", "ISRC3DEC3"), + ARIZONA_MUX_ROUTES("ISRC3DEC4", "ISRC3DEC4"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, + { "DRC2 Signal Activity", NULL, "DRC2L" }, + { "DRC2 Signal Activity", NULL, "DRC2R" }, +}; + +static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct cs47l24_priv *cs47l24 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case CS47L24_FLL1: + return arizona_set_fll(&cs47l24->fll[0], source, Fref, Fout); + case CS47L24_FLL2: + return arizona_set_fll(&cs47l24->fll[1], source, Fref, Fout); + case CS47L24_FLL1_REFCLK: + return arizona_set_fll_refclk(&cs47l24->fll[0], source, Fref, + Fout); + case CS47L24_FLL2_REFCLK: + return arizona_set_fll_refclk(&cs47l24->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000 + +#define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver cs47l24_dai[] = { + { + .name = "cs47l24-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l24-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l24-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, +}; + +static int cs47l24_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + priv->core.arizona->dapm = dapm; + + arizona_init_spk(codec); + arizona_init_gpio(codec); + arizona_init_mono(codec); + + ret = wm_adsp2_codec_probe(&priv->core.adsp[1], codec); + if (ret) + goto err_adsp2_codec_probe; + + ret = wm_adsp2_codec_probe(&priv->core.adsp[2], codec); + if (ret) + goto err_adsp2_codec_probe; + + ret = snd_soc_add_codec_controls(codec, + &arizona_adsp2_rate_controls[1], 2); + if (ret) + goto err_adsp2_codec_probe; + + snd_soc_dapm_disable_pin(dapm, "HAPTICS"); + + return 0; + +err_adsp2_codec_probe: + wm_adsp2_codec_remove(&priv->core.adsp[1], codec); + wm_adsp2_codec_remove(&priv->core.adsp[2], codec); + + return ret; +} + +static int cs47l24_codec_remove(struct snd_soc_codec *codec) +{ + struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec); + + + wm_adsp2_codec_remove(&priv->core.adsp[1], codec); + wm_adsp2_codec_remove(&priv->core.adsp[2], codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define CS47L24_DIG_VU 0x0200 + +static unsigned int cs47l24_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, +}; + +static struct regmap *cs47l24_get_regmap(struct device *dev) +{ + struct cs47l24_priv *priv = dev_get_drvdata(dev); + + return priv->core.arizona->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = { + .probe = cs47l24_codec_probe, + .remove = cs47l24_codec_remove, + .get_regmap = cs47l24_get_regmap, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = cs47l24_set_fll, + + .controls = cs47l24_snd_controls, + .num_controls = ARRAY_SIZE(cs47l24_snd_controls), + .dapm_widgets = cs47l24_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs47l24_dapm_widgets), + .dapm_routes = cs47l24_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs47l24_dapm_routes), +}; + +static int cs47l24_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct cs47l24_priv *cs47l24; + int i, ret; + + BUILD_BUG_ON(ARRAY_SIZE(cs47l24_dai) > ARIZONA_MAX_DAI); + + cs47l24 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l24_priv), + GFP_KERNEL); + if (!cs47l24) + return -ENOMEM; + + platform_set_drvdata(pdev, cs47l24); + + cs47l24->core.arizona = arizona; + cs47l24->core.num_inputs = 4; + + for (i = 1; i <= 2; i++) { + cs47l24->core.adsp[i].part = "cs47l24"; + cs47l24->core.adsp[i].num = i + 1; + cs47l24->core.adsp[i].type = WMFW_ADSP2; + cs47l24->core.adsp[i].dev = arizona->dev; + cs47l24->core.adsp[i].regmap = arizona->regmap; + + cs47l24->core.adsp[i].base = ARIZONA_DSP1_CONTROL_1 + + (0x100 * i); + cs47l24->core.adsp[i].mem = cs47l24_dsp_regions[i - 1]; + cs47l24->core.adsp[i].num_mems = + ARRAY_SIZE(cs47l24_dsp2_regions); + + ret = wm_adsp2_init(&cs47l24->core.adsp[i]); + if (ret != 0) + return ret; + } + + for (i = 0; i < ARRAY_SIZE(cs47l24->fll); i++) + cs47l24->fll[i].vco_mult = 3; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &cs47l24->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &cs47l24->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(cs47l24_dai); i++) + arizona_init_dai(&cs47l24->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(cs47l24_digital_vu); i++) + regmap_update_bits(arizona->regmap, cs47l24_digital_vu[i], + CS47L24_DIG_VU, CS47L24_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, + cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); +} + +static int cs47l24_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver cs47l24_codec_driver = { + .driver = { + .name = "cs47l24-codec", + }, + .probe = cs47l24_probe, + .remove = cs47l24_remove, +}; + +module_platform_driver(cs47l24_codec_driver); + +MODULE_DESCRIPTION("ASoC CS47L24 driver"); +MODULE_AUTHOR("Richard Fitzgerald "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cs47l24-codec"); diff --git a/sound/soc/codecs/cs47l24.h b/sound/soc/codecs/cs47l24.h new file mode 100644 index 000000000000..77ab2b77b2e6 --- /dev/null +++ b/sound/soc/codecs/cs47l24.h @@ -0,0 +1,23 @@ +/* + * cs47l24.h -- ALSA SoC Audio driver for Cirrus Logic CS47L24 + * + * Copyright 2015 Cirrus Logic Inc. + * + * Author: Richard Fitzgerald + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _CS47L24_H +#define _CS47L24_H + +#include "arizona.h" + +#define CS47L24_FLL1 1 +#define CS47L24_FLL2 2 +#define CS47L24_FLL1_REFCLK 3 +#define CS47L24_FLL2_REFCLK 4 + +#endif From a3af0c65836e714fa71dcaa0a81f6db83a212faa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Nov 2015 04:51:21 +0000 Subject: [PATCH 02/23] ASoC: ak4613: add single-end optional property for IN/OUT pins ak4613 IN/OUT pin can be selected as differential/single-end. Default is differential, because it is register default settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ak4613.txt | 10 +++++++ sound/soc/codecs/ak4613.c | 29 +++++++++++++++++++ 2 files changed, 39 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt index 15a919522b42..3cf63e7f8e77 100644 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -7,6 +7,16 @@ Required properties: - compatible : "asahi-kasei,ak4613" - reg : The chip select number on the I2C bus +Optional properties: +- ak4613,in1-single-end : Boolean. Indicate input / output pins are single-ended. +- ak4613,in2-single-end rather than differential. +- ak4613,out1-single-end +- ak4613,out2-single-end +- ak4613,out3-single-end +- ak4613,out4-single-end +- ak4613,out5-single-end +- ak4613,out6-single-end + Example: &i2c { diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 07a266460ec3..394c10ff049e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -79,6 +79,8 @@ struct ak4613_priv { unsigned int fmt; u8 fmt_ctrl; + u8 oc; + u8 ic; int cnt; }; @@ -343,6 +345,9 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); snd_soc_write(codec, CTRL2, ctrl2); + snd_soc_write(codec, ICTRL, priv->ic); + snd_soc_write(codec, OCTRL, priv->oc); + hw_params_end: if (ret < 0) dev_warn(dev, "unsupported data width/format combination\n"); @@ -431,6 +436,28 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), }; +static void ak4613_parse_of(struct ak4613_priv *priv, + struct device *dev) +{ + struct device_node *np = dev->of_node; + char prop[32]; + int i; + + /* Input 1 - 2 */ + for (i = 0; i < 2; i++) { + snprintf(prop, sizeof(prop), "ak4613,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), "ak4613,out%d-single-end", i + 1); + if (!of_get_property(np, prop, NULL)) + priv->oc |= 1 << i; + } +} + static int ak4613_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -458,6 +485,8 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; + ak4613_parse_of(priv, dev); + priv->fmt_ctrl = NO_FMT; priv->cnt = 0; From 35299f1779dbdcb61af4305904963b5bc9276eb9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Nov 2015 06:01:29 +0000 Subject: [PATCH 03/23] ASoC: ak4613: tidyup CTRL1 value selection method Current CTRL1 selection method didn't care about simultaneous playback / capture. This patch tidyup it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 94 +++++++++++++++++++++++---------------- 1 file changed, 56 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 394c10ff049e..dab127603ff6 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -74,16 +74,6 @@ #define DFS_DOUBLE_SPEED (1 << 2) #define DFS_QUAD_SPEED (2 << 2) -struct ak4613_priv { - struct mutex lock; - - unsigned int fmt; - u8 fmt_ctrl; - u8 oc; - u8 ic; - int cnt; -}; - struct ak4613_formats { unsigned int width; unsigned int fmt; @@ -94,6 +84,16 @@ struct ak4613_interface { struct ak4613_formats playback; }; +struct ak4613_priv { + struct mutex lock; + const struct ak4613_interface *iface; + + unsigned int fmt; + u8 oc; + u8 ic; + int cnt; +}; + /* * Playback Volume * @@ -128,7 +128,7 @@ static const struct reg_default ak4613_reg[] = { { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, }; -#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE_TO_VAL(fmts) ((fmts - ak4613_iface) << 3) #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ @@ -242,7 +242,7 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, priv->cnt = 0; } if (!priv->cnt) - priv->fmt_ctrl = NO_FMT; + priv->iface = NULL; mutex_unlock(&priv->lock); } @@ -267,13 +267,35 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, + int is_play, + unsigned int fmt, unsigned int width) +{ + const struct ak4613_formats *fmts; + + fmts = (is_play) ? &iface->playback : &iface->capture; + + if (fmts->fmt != fmt) + return false; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + return false; + } else { + if (fmts->width < width) + return false; + } + + return true; +} + 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_codec *codec = dai->codec; struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); - const struct ak4613_formats *fmts; + const struct ak4613_interface *iface; struct device *dev = codec->dev; unsigned int width = params_width(params); unsigned int fmt = priv->fmt; @@ -307,33 +329,27 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, * It doesn't support TDM at this point */ fmt_ctrl = NO_FMT; - for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { - fmts = (is_play) ? &ak4613_iface[i].playback : - &ak4613_iface[i].capture; - - if (fmts->fmt != fmt) - continue; - - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - continue; - } else { - if (fmts->width < width) - continue; - } - - fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); - break; - } - ret = -EINVAL; - if (fmt_ctrl == NO_FMT) - goto hw_params_end; + iface = NULL; mutex_lock(&priv->lock); - if ((priv->fmt_ctrl == NO_FMT) || - (priv->fmt_ctrl == fmt_ctrl)) { - priv->fmt_ctrl = fmt_ctrl; + if (priv->iface) { + if (ak4613_dai_fmt_matching(priv->iface, is_play, fmt, width)) + iface = priv->iface; + } else { + for (i = ARRAY_SIZE(ak4613_iface); i >= 0; i--) { + if (!ak4613_dai_fmt_matching(ak4613_iface + i, + is_play, + fmt, width)) + continue; + iface = ak4613_iface + i; + break; + } + } + + if ((priv->iface == NULL) || + (priv->iface == iface)) { + priv->iface = iface; priv->cnt++; ret = 0; } @@ -342,6 +358,8 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto hw_params_end; + fmt_ctrl = AUDIO_IFACE_TO_VAL(iface); + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); snd_soc_write(codec, CTRL2, ctrl2); @@ -487,7 +505,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, ak4613_parse_of(priv, dev); - priv->fmt_ctrl = NO_FMT; + priv->iface = NULL; priv->cnt = 0; mutex_init(&priv->lock); From b323dd30718e2055adb5534e52f685a57c119c18 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 17 Nov 2015 08:28:56 +0000 Subject: [PATCH 04/23] ASoC: ak4613: don't overwrite CTRL2 register Current code set DFS settings on CTRL2 register, but it overwrite default settings. This patch fixup it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index dab127603ff6..62c08a6395af 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -70,6 +70,7 @@ #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) @@ -361,7 +362,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, fmt_ctrl = AUDIO_IFACE_TO_VAL(iface); snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); - snd_soc_write(codec, CTRL2, ctrl2); + snd_soc_update_bits(codec, CTRL2, DFS_MASK, ctrl2); snd_soc_write(codec, ICTRL, priv->ic); snd_soc_write(codec, OCTRL, priv->oc); From 9bf5c3d11f1fbaf43399d189f05fb20ceb46ee5d Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Wed, 11 Nov 2015 13:12:51 +0100 Subject: [PATCH 05/23] ASoC: ac97: add gpio chip The AC97 specification provides a guide for 16 GPIOs in the codecs. If the gpiolib is compiled in the kernel, declare a gpio chip. This was tested with a pxa27x board (mioa701) and a wm9713 codec. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- include/sound/ac97_codec.h | 3 + sound/soc/soc-ac97.c | 125 +++++++++++++++++++++++++++++++++++++ 2 files changed, 128 insertions(+) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 74bc85473b58..15aa5f07c955 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -417,11 +417,13 @@ #define AC97_RATES_MIC_ADC 4 #define AC97_RATES_SPDIF 5 +#define AC97_NUM_GPIOS 16 /* * */ struct snd_ac97; +struct snd_ac97_gpio_priv; struct snd_pcm_chmap; struct snd_ac97_build_ops { @@ -529,6 +531,7 @@ struct snd_ac97 { struct delayed_work power_work; #endif struct device dev; + struct snd_ac97_gpio_priv *gpio_priv; struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */ }; diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index d40efc9fe0a9..ae563e379a72 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -38,6 +39,14 @@ struct snd_ac97_reset_cfg { int gpio_reset; }; +struct snd_ac97_gpio_priv { +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif + unsigned int gpios_set; + struct snd_soc_codec *codec; +}; + static struct snd_ac97_bus soc_ac97_bus = { .ops = NULL, /* Gets initialized in snd_soc_set_ac97_ops() */ }; @@ -47,6 +56,117 @@ static void soc_ac97_device_release(struct device *dev) kfree(to_ac97_t(dev)); } +#ifdef CONFIG_GPIOLIB +static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) +{ + struct snd_ac97_gpio_priv *gpio_priv = + container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + + return gpio_priv->codec; +} + +static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + if (offset >= AC97_NUM_GPIOS) + return -EINVAL; + + return 0; +} + +static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, + unsigned offset) +{ + struct snd_soc_codec *codec = gpio_to_codec(chip); + + dev_dbg(codec->dev, "set gpio %d to output\n", offset); + return snd_soc_update_bits(codec, AC97_GPIO_CFG, + 1 << offset, 1 << offset); +} + +static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct snd_soc_codec *codec = gpio_to_codec(chip); + int ret; + + ret = snd_soc_read(codec, AC97_GPIO_STATUS); + dev_dbg(codec->dev, "get gpio %d : %d\n", offset, + ret < 0 ? ret : ret & (1 << offset)); + + return ret < 0 ? ret : ret & (1 << offset); +} + +static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, + int value) +{ + struct snd_ac97_gpio_priv *gpio_priv = + container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_soc_codec *codec = gpio_to_codec(chip); + + gpio_priv->gpios_set &= ~(1 << offset); + gpio_priv->gpios_set |= (!!value) << offset; + snd_soc_write(codec, AC97_GPIO_STATUS, gpio_priv->gpios_set); + dev_dbg(codec->dev, "set gpio %d to %d\n", offset, !!value); +} + +static int snd_soc_ac97_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct snd_soc_codec *codec = gpio_to_codec(chip); + + dev_dbg(codec->dev, "set gpio %d to output\n", offset); + snd_soc_ac97_gpio_set(chip, offset, value); + return snd_soc_update_bits(codec, AC97_GPIO_CFG, 1 << offset, 0); +} + +static struct gpio_chip snd_soc_ac97_gpio_chip = { + .label = "snd_soc_ac97", + .owner = THIS_MODULE, + .request = snd_soc_ac97_gpio_request, + .direction_input = snd_soc_ac97_gpio_direction_in, + .get = snd_soc_ac97_gpio_get, + .direction_output = snd_soc_ac97_gpio_direction_out, + .set = snd_soc_ac97_gpio_set, + .can_sleep = 1, +}; + +static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, + struct snd_soc_codec *codec) +{ + struct snd_ac97_gpio_priv *gpio_priv; + int ret; + + gpio_priv = devm_kzalloc(codec->dev, sizeof(*gpio_priv), GFP_KERNEL); + if (!gpio_priv) + return -ENOMEM; + ac97->gpio_priv = gpio_priv; + gpio_priv->codec = codec; + gpio_priv->gpio_chip = snd_soc_ac97_gpio_chip; + gpio_priv->gpio_chip.ngpio = AC97_NUM_GPIOS; + gpio_priv->gpio_chip.dev = codec->dev; + gpio_priv->gpio_chip.base = -1; + + ret = gpiochip_add(&gpio_priv->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + return ret; +} + +static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) +{ + gpiochip_remove(&ac97->gpio_priv->gpio_chip); +} +#else +static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, + struct snd_soc_codec *codec) +{ + return 0; +} + +static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) +{ +} +#endif + /** * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device * @codec: The CODEC for which to create the AC'97 device @@ -119,6 +239,10 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, if (ret) goto err_put_device; + ret = snd_soc_ac97_init_gpio(ac97, codec); + if (ret) + goto err_put_device; + return ac97; err_put_device: @@ -135,6 +259,7 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); */ void snd_soc_free_ac97_codec(struct snd_ac97 *ac97) { + snd_soc_ac97_free_gpio(ac97); device_del(&ac97->dev); ac97->bus = NULL; put_device(&ac97->dev); From 5547ba616b964de05ba48ec4d529ed1ac22a4326 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Nov 2015 04:22:14 +0000 Subject: [PATCH 06/23] ASoC: ak4613: tidyup vendor prefix from ak4613 to asahi-kasei a3af0c65("ASoC: ak4613: add single-end optional property for IN/OUT pins") added IN/OUT pin single-end optional property, but it used "ak4613" as vendor prefix. This patch fixup to asahi-kasei. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ak4613.txt | 16 ++++++++-------- sound/soc/codecs/ak4613.c | 4 ++-- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt index 3cf63e7f8e77..1783f9ef0930 100644 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -8,14 +8,14 @@ Required properties: - reg : The chip select number on the I2C bus Optional properties: -- ak4613,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- ak4613,in2-single-end rather than differential. -- ak4613,out1-single-end -- ak4613,out2-single-end -- ak4613,out3-single-end -- ak4613,out4-single-end -- ak4613,out5-single-end -- ak4613,out6-single-end +- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. +- asahi-kasei,in2-single-end rather than differential. +- asahi-kasei,out1-single-end +- asahi-kasei,out2-single-end +- asahi-kasei,out3-single-end +- asahi-kasei,out4-single-end +- asahi-kasei,out5-single-end +- asahi-kasei,out6-single-end Example: diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 62c08a6395af..647f69de6baa 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -464,14 +464,14 @@ static void ak4613_parse_of(struct ak4613_priv *priv, /* Input 1 - 2 */ for (i = 0; i < 2; i++) { - snprintf(prop, sizeof(prop), "ak4613,in%d-single-end", i + 1); + 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), "ak4613,out%d-single-end", i + 1); + snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1); if (!of_get_property(np, prop, NULL)) priv->oc |= 1 << i; } From 04d1300fd3f5c070a9abe391860d29dfcda89c87 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 26 Nov 2015 14:01:52 +0000 Subject: [PATCH 07/23] ASoC: wm_adsp: Expand the list of available firmwares Expand the list of available firmware names to include a good selection of generic uses for the DSP cores. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 47 +++++++++++++++++++++++++++----------- 1 file changed, 34 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0bb415a28723..905ae993440b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -201,27 +201,48 @@ static void wm_adsp_buf_free(struct list_head *list) } } -#define WM_ADSP_NUM_FW 4 +#define WM_ADSP_FW_MBC_VSS 0 +#define WM_ADSP_FW_HIFI 1 +#define WM_ADSP_FW_TX 2 +#define WM_ADSP_FW_TX_SPK 3 +#define WM_ADSP_FW_RX 4 +#define WM_ADSP_FW_RX_ANC 5 +#define WM_ADSP_FW_CTRL 6 +#define WM_ADSP_FW_ASR 7 +#define WM_ADSP_FW_TRACE 8 +#define WM_ADSP_FW_SPK_PROT 9 +#define WM_ADSP_FW_MISC 10 -#define WM_ADSP_FW_MBC_VSS 0 -#define WM_ADSP_FW_TX 1 -#define WM_ADSP_FW_TX_SPK 2 -#define WM_ADSP_FW_RX_ANC 3 +#define WM_ADSP_NUM_FW 11 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", - [WM_ADSP_FW_TX] = "Tx", - [WM_ADSP_FW_TX_SPK] = "Tx Speaker", - [WM_ADSP_FW_RX_ANC] = "Rx ANC", + [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", + [WM_ADSP_FW_HIFI] = "MasterHiFi", + [WM_ADSP_FW_TX] = "Tx", + [WM_ADSP_FW_TX_SPK] = "Tx Speaker", + [WM_ADSP_FW_RX] = "Rx", + [WM_ADSP_FW_RX_ANC] = "Rx ANC", + [WM_ADSP_FW_CTRL] = "Voice Ctrl", + [WM_ADSP_FW_ASR] = "ASR Assist", + [WM_ADSP_FW_TRACE] = "Dbg Trace", + [WM_ADSP_FW_SPK_PROT] = "Protection", + [WM_ADSP_FW_MISC] = "Misc", }; static struct { const char *file; } wm_adsp_fw[WM_ADSP_NUM_FW] = { - [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, - [WM_ADSP_FW_TX] = { .file = "tx" }, - [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, - [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, + [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, + [WM_ADSP_FW_HIFI] = { .file = "hifi" }, + [WM_ADSP_FW_TX] = { .file = "tx" }, + [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, + [WM_ADSP_FW_RX] = { .file = "rx" }, + [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, + [WM_ADSP_FW_CTRL] = { .file = "ctrl" }, + [WM_ADSP_FW_ASR] = { .file = "asr" }, + [WM_ADSP_FW_TRACE] = { .file = "trace" }, + [WM_ADSP_FW_SPK_PROT] = { .file = "spk-prot" }, + [WM_ADSP_FW_MISC] = { .file = "misc" }, }; struct wm_coeff_ctl_ops { From 078e71838cdff1c2a1a33e65459954adda9a4641 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 8 Dec 2015 16:08:26 +0000 Subject: [PATCH 08/23] ASoC: wm_adsp: Replace debugfs lock with more general DSP power lock Most events around the DSP just need to be locked to ensure that the DSP can't change power state whilst they are happening. This includes the debugfs entries and this will make sorting the rest of the locking simpler. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 75 +++++++++++++++++++++----------------- sound/soc/codecs/wm_adsp.h | 3 +- 2 files changed, 44 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 905ae993440b..19f05933de54 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -275,30 +275,24 @@ static void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, const char *s) { char *tmp = kasprintf(GFP_KERNEL, "%s\n", s); - mutex_lock(&dsp->debugfs_lock); kfree(dsp->wmfw_file_name); dsp->wmfw_file_name = tmp; - mutex_unlock(&dsp->debugfs_lock); } static void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, const char *s) { char *tmp = kasprintf(GFP_KERNEL, "%s\n", s); - mutex_lock(&dsp->debugfs_lock); kfree(dsp->bin_file_name); dsp->bin_file_name = tmp; - mutex_unlock(&dsp->debugfs_lock); } static void wm_adsp_debugfs_clear(struct wm_adsp *dsp) { - mutex_lock(&dsp->debugfs_lock); kfree(dsp->wmfw_file_name); kfree(dsp->bin_file_name); dsp->wmfw_file_name = NULL; dsp->bin_file_name = NULL; - mutex_unlock(&dsp->debugfs_lock); } static ssize_t wm_adsp_debugfs_wmfw_read(struct file *file, @@ -308,7 +302,7 @@ static ssize_t wm_adsp_debugfs_wmfw_read(struct file *file, struct wm_adsp *dsp = file->private_data; ssize_t ret; - mutex_lock(&dsp->debugfs_lock); + mutex_lock(&dsp->pwr_lock); if (!dsp->wmfw_file_name || !dsp->running) ret = 0; @@ -317,7 +311,7 @@ static ssize_t wm_adsp_debugfs_wmfw_read(struct file *file, dsp->wmfw_file_name, strlen(dsp->wmfw_file_name)); - mutex_unlock(&dsp->debugfs_lock); + mutex_unlock(&dsp->pwr_lock); return ret; } @@ -328,7 +322,7 @@ static ssize_t wm_adsp_debugfs_bin_read(struct file *file, struct wm_adsp *dsp = file->private_data; ssize_t ret; - mutex_lock(&dsp->debugfs_lock); + mutex_lock(&dsp->pwr_lock); if (!dsp->bin_file_name || !dsp->running) ret = 0; @@ -337,7 +331,7 @@ static ssize_t wm_adsp_debugfs_bin_read(struct file *file, dsp->bin_file_name, strlen(dsp->bin_file_name)); - mutex_unlock(&dsp->debugfs_lock); + mutex_unlock(&dsp->pwr_lock); return ret; } @@ -1799,9 +1793,8 @@ int wm_adsp1_init(struct wm_adsp *dsp) { INIT_LIST_HEAD(&dsp->alg_regions); -#ifdef CONFIG_DEBUG_FS - mutex_init(&dsp->debugfs_lock); -#endif + mutex_init(&dsp->pwr_lock); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp1_init); @@ -1820,6 +1813,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, dsp->card = codec->component.card; + mutex_lock(&dsp->pwr_lock); + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1834,7 +1829,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); - return ret; + goto err_mutex; } val = (val & dsp->sysclk_mask) @@ -1846,31 +1841,31 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); - return ret; + goto err_mutex; } } ret = wm_adsp_load(dsp); if (ret != 0) - goto err; + goto err_ena; ret = wm_adsp1_setup_algs(dsp); if (ret != 0) - goto err; + goto err_ena; ret = wm_adsp_load_coeff(dsp); if (ret != 0) - goto err; + goto err_ena; /* Initialize caches for enabled and unset controls */ ret = wm_coeff_init_control_caches(dsp); if (ret != 0) - goto err; + goto err_ena; /* Sync set controls */ ret = wm_coeff_sync_controls(dsp); if (ret != 0) - goto err; + goto err_ena; /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1905,11 +1900,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, break; } + mutex_unlock(&dsp->pwr_lock); + return 0; -err: +err_ena: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); +err_mutex: + mutex_unlock(&dsp->pwr_lock); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp1_event); @@ -1955,6 +1955,8 @@ static void wm_adsp2_boot_work(struct work_struct *work) int ret; unsigned int val; + mutex_lock(&dsp->pwr_lock); + /* * For simplicity set the DSP clock rate to be the * SYSCLK rate rather than making it configurable. @@ -1962,7 +1964,7 @@ static void wm_adsp2_boot_work(struct work_struct *work) ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); - return; + goto err_mutex; } val = (val & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; @@ -1972,42 +1974,46 @@ static void wm_adsp2_boot_work(struct work_struct *work) ADSP2_CLK_SEL_MASK, val); if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); - return; + goto err_mutex; } ret = wm_adsp2_ena(dsp); if (ret != 0) - return; + goto err_mutex; ret = wm_adsp_load(dsp); if (ret != 0) - goto err; + goto err_ena; ret = wm_adsp2_setup_algs(dsp); if (ret != 0) - goto err; + goto err_ena; ret = wm_adsp_load_coeff(dsp); if (ret != 0) - goto err; + goto err_ena; /* Initialize caches for enabled and unset controls */ ret = wm_coeff_init_control_caches(dsp); if (ret != 0) - goto err; + goto err_ena; /* Sync set controls */ ret = wm_coeff_sync_controls(dsp); if (ret != 0) - goto err; + goto err_ena; dsp->running = true; + mutex_unlock(&dsp->pwr_lock); + return; -err: +err_ena: regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); +err_mutex: + mutex_unlock(&dsp->pwr_lock); } int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, @@ -2060,6 +2066,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, /* Log firmware state, it can be useful for analysis */ wm_adsp2_show_fw_status(dsp); + mutex_lock(&dsp->pwr_lock); + wm_adsp_debugfs_clear(dsp); dsp->fw_id = 0; @@ -2086,6 +2094,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, kfree(alg_region); } + mutex_unlock(&dsp->pwr_lock); + adsp_dbg(dsp, "Shutdown complete\n"); break; @@ -2138,9 +2148,8 @@ int wm_adsp2_init(struct wm_adsp *dsp) INIT_LIST_HEAD(&dsp->ctl_list); INIT_WORK(&dsp->boot_work, wm_adsp2_boot_work); -#ifdef CONFIG_DEBUG_FS - mutex_init(&dsp->debugfs_lock); -#endif + mutex_init(&dsp->pwr_lock); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 2d117cf0e953..93764139313b 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -59,9 +59,10 @@ struct wm_adsp { struct work_struct boot_work; + struct mutex pwr_lock; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_root; - struct mutex debugfs_lock; char *wmfw_file_name; char *bin_file_name; #endif From d27c5e155c69a4c45e9833fbf66aa580dcd01624 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 8 Dec 2015 16:08:28 +0000 Subject: [PATCH 09/23] ASoC: wm_adsp: Add power lock for firmware change control We should hold the DSP power lock whilst changing the firmware since we need to check if it is running first. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 19f05933de54..fd85a8cc7234 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -451,6 +451,7 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + int ret = 0; if (ucontrol->value.integer.value[0] == dsp[e->shift_l].fw) return 0; @@ -458,12 +459,16 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] >= WM_ADSP_NUM_FW) return -EINVAL; + mutex_lock(&dsp[e->shift_l].pwr_lock); + if (dsp[e->shift_l].running) - return -EBUSY; + ret = -EBUSY; + else + dsp[e->shift_l].fw = ucontrol->value.integer.value[0]; - dsp[e->shift_l].fw = ucontrol->value.integer.value[0]; + mutex_unlock(&dsp[e->shift_l].pwr_lock); - return 0; + return ret; } static const struct soc_enum wm_adsp_fw_enum[] = { From 7585a5b0ab5511376f032e421f7de72fe7e160d5 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 8 Dec 2015 16:08:25 +0000 Subject: [PATCH 10/23] ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++-------------- sound/soc/codecs/wm_adsp.h | 4 ++-- 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fd85a8cc7234..3a314f2a3f61 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -543,10 +543,10 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } -static int wm_coeff_info(struct snd_kcontrol *kcontrol, +static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -592,10 +592,10 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_put(struct snd_kcontrol *kcontrol, +static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; memcpy(ctl->cache, p, ctl->len); @@ -646,10 +646,10 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kcontrol, +static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { @@ -828,8 +828,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, break; } - list_for_each_entry(ctl, &dsp->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; @@ -1108,7 +1107,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - header = (void*)&firmware->data[0]; + header = (void *)&firmware->data[0]; if (memcmp(&header->magic[0], "WMFW", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); @@ -1188,7 +1187,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) offset = le32_to_cpu(region->offset) & 0xffffff; type = be32_to_cpu(region->type) & 0xff; mem = wm_adsp_find_region(dsp, type); - + switch (type) { case WMFW_NAME_TEXT: region_name = "Firmware name"; @@ -1645,7 +1644,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) goto out_fw; } - hdr = (void*)&firmware->data[0]; + hdr = (void *)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); goto out_fw; @@ -1671,7 +1670,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && pos - firmware->size > sizeof(*blk)) { - blk = (void*)(&firmware->data[pos]); + blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); offset = le16_to_cpu(blk->offset); @@ -1814,7 +1813,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; - int val; + unsigned int val; dsp->card = codec->component.card; @@ -1829,7 +1828,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, * For simplicity set the DSP clock rate to be the * SYSCLK rate rather than making it configurable. */ - if(dsp->sysclk_reg) { + if (dsp->sysclk_reg) { ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 93764139313b..d2a8c78ed50b 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -45,8 +45,8 @@ struct wm_adsp { struct list_head alg_regions; - int fw_id; - int fw_id_version; + unsigned int fw_id; + unsigned int fw_id_version; const struct wm_adsp_region *mem; int num_mems; From 168d10e74c4efd945a37adeb134f096505e62b49 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 8 Dec 2015 16:08:27 +0000 Subject: [PATCH 11/23] ASoC: wm_adsp: Add locking to DSP firmware controls Locking is currently missing from the DSP firmware controls, which can lead to some race conditions if the controls are accessed as the DSP powers up or down. This patch adds them to the new power lock. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3a314f2a3f61..b083642718f0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -597,14 +597,19 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, { struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); memcpy(ctl->cache, p, ctl->len); ctl->set = 1; - if (!ctl->enabled) - return 0; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, p, ctl->len); - return wm_coeff_write_control(ctl, p, ctl->len); + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; } static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, @@ -651,17 +656,22 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, { struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { if (ctl->enabled) - return wm_coeff_read_control(ctl, p, ctl->len); + ret = wm_coeff_read_control(ctl, p, ctl->len); else - return -EPERM; + ret = -EPERM; + } else { + memcpy(p, ctl->cache, ctl->len); } - memcpy(p, ctl->cache, ctl->len); + mutex_unlock(&ctl->dsp->pwr_lock); - return 0; + return ret; } struct wmfw_ctl_work { From 32e69bad8ed9ed4e1bf1fd8217b61d5f87996253 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Fri, 11 Dec 2015 11:07:37 +0800 Subject: [PATCH 12/23] ASoC: Atmel: ClassD: unregister codec when error occurs Add code to unregister codec in probe function, when the error occurs after the codec is registered. Signed-off-by: Songjun Wu Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-classd.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 8276675730ef..f3ffb39bfe27 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -636,8 +636,10 @@ static int atmel_classd_probe(struct platform_device *pdev) /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); - if (!card) - return -ENOMEM; + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } snd_soc_card_set_drvdata(card, dd); platform_set_drvdata(pdev, card); @@ -645,16 +647,20 @@ static int atmel_classd_probe(struct platform_device *pdev) ret = atmel_classd_asoc_card_init(dev, card); if (ret) { dev_err(dev, "failed to init sound card\n"); - return ret; + goto unregister_codec; } ret = devm_snd_soc_register_card(dev, card); if (ret) { dev_err(dev, "failed to register sound card: %d\n", ret); - return ret; + goto unregister_codec; } return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; } static int atmel_classd_remove(struct platform_device *pdev) From bc1765d6e81a70be36a7290bcc829a7714101bbb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 17 Dec 2015 10:05:59 +0000 Subject: [PATCH 13/23] ASoC: wm_adsp: Mimic legacy behaviour of reading controls when DSP is on Older firmwares don't specify access flags for the controls, unfortunately the usage of some of these firmware relies on being able to read back values from the DSP. The current control code will only do this for volatile controls. This patch will read the control from the hardware if no flags are specified and the control is currently enabled, which should cover these legacy use-cases. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b083642718f0..d1e0826c7db2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -666,6 +666,9 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, else ret = -EPERM; } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + memcpy(p, ctl->cache, ctl->len); } From a7664ab29af7d7eca57ae525b5063f71fa006ff4 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Thu, 17 Dec 2015 17:49:59 +0800 Subject: [PATCH 14/23] ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller Add driver for the Pulse Density Modulation Interface Controller. It comes with digitallly controlled gain, a High-Pass and a SINCC filter. Signed-off-by: Songjun Wu Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 738 ++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 ++++ 4 files changed, 829 insertions(+) create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 2d30464b81ce..06e099e802df 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -68,4 +68,13 @@ config SND_ATMEL_SOC_CLASSD help Say Y if you want to add support for Atmel ASoC driver for boards using CLASSD. + +config SND_ATMEL_SOC_PDMIC + tristate "Atmel ASoC driver for boards using PDMIC" + depends on OF && (ARCH_AT91 || COMPILE_TEST) + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y if you want to add support for Atmel ASoC driver for boards using + PDMIC. endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f6f7db428216..a2b127bd9c87 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -12,8 +12,10 @@ snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o snd-atmel-soc-wm8904-objs := atmel_wm8904.o snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o snd-atmel-soc-classd-objs := atmel-classd.o +snd-atmel-soc-pdmic-objs := atmel-pdmic.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o +obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c new file mode 100644 index 000000000000..aee4787a0b89 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.c @@ -0,0 +1,738 @@ +/* Atmel PDMIC driver + * + * Copyright (C) 2015 Atmel + * + * Author: Songjun Wu + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 or later + * as published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "atmel-pdmic.h" + +struct atmel_pdmic_pdata { + u32 mic_min_freq; + u32 mic_max_freq; + s32 mic_offset; + const char *card_name; +}; + +struct atmel_pdmic { + dma_addr_t phy_base; + struct regmap *regmap; + struct clk *pclk; + struct clk *gclk; + int irq; + struct snd_pcm_substream *substream; + const struct atmel_pdmic_pdata *pdata; +}; + +static const struct of_device_id atmel_pdmic_of_match[] = { + { + .compatible = "atmel,sama5d2-pdmic", + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, atmel_pdmic_of_match); + +#define PDMIC_OFFSET_MAX_VAL S16_MAX +#define PDMIC_OFFSET_MIN_VAL S16_MIN + +static struct atmel_pdmic_pdata *atmel_pdmic_dt_init(struct device *dev) +{ + struct device_node *np = dev->of_node; + struct atmel_pdmic_pdata *pdata; + + if (!np) { + dev_err(dev, "device node not found\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + if (of_property_read_string(np, "atmel,model", &pdata->card_name)) + pdata->card_name = "PDMIC"; + + if (of_property_read_u32(np, "atmel,mic-min-freq", + &pdata->mic_min_freq)) { + dev_err(dev, "failed to get mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_u32(np, "atmel,mic-max-freq", + &pdata->mic_max_freq)) { + dev_err(dev, "failed to get mic-max-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (pdata->mic_max_freq < pdata->mic_min_freq) { + dev_err(dev, + "mic-max-freq should not less than mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_s32(np, "atmel,mic-offset", &pdata->mic_offset)) + pdata->mic_offset = 0; + + if (pdata->mic_offset > PDMIC_OFFSET_MAX_VAL) { + dev_warn(dev, + "mic-offset value %d is larger than the max value %d, the max value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MAX_VAL); + pdata->mic_offset = PDMIC_OFFSET_MAX_VAL; + } else if (pdata->mic_offset < PDMIC_OFFSET_MIN_VAL) { + dev_warn(dev, + "mic-offset value %d is less than the min value %d, the min value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MIN_VAL); + pdata->mic_offset = PDMIC_OFFSET_MIN_VAL; + } + + return pdata; +} + +/* cpu dai component */ +static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = clk_prepare_enable(dd->gclk); + if (ret) + return ret; + + ret = clk_prepare_enable(dd->pclk); + if (ret) + return ret; + + /* Clear all bits in the Control Register(PDMIC_CR) */ + regmap_write(dd->regmap, PDMIC_CR, 0); + + dd->substream = substream; + + /* Enable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IER, PDMIC_IER_OVRE); + + return 0; +} + +static void atmel_pdmic_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + + /* Disable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IDR, PDMIC_IDR_OVRE); + + clk_disable_unprepare(dd->gclk); + clk_disable_unprepare(dd->pclk); +} + +static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + /* Clean the PDMIC Converted Data Register */ + return regmap_read(dd->regmap, PDMIC_CDR, &val); +} + +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, +}; + +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS,}, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", +}; + +/* platform */ +#define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) +#define ATMEL_PDMIC_PREALLOC_BUF_SIZE ATMEL_PDMIC_MAX_BUF_SIZE + +static const struct snd_pcm_hardware atmel_pdmic_hw = { + .info = SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE, + .formats = ATMEL_PDMIC_FORMATS, + .buffer_bytes_max = ATMEL_PDMIC_MAX_BUF_SIZE, + .period_bytes_min = 256, + .period_bytes_max = 32 * 1024, + .periods_min = 2, + .periods_max = 256, +}; + +static int +atmel_pdmic_platform_configure_dma(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = snd_hwparams_to_dma_slave_config(substream, params, + slave_config); + if (ret) { + dev_err(rtd->platform->dev, + "hw params to dma slave configure failed\n"); + return ret; + } + + slave_config->src_addr = dd->phy_base + PDMIC_CDR; + slave_config->src_maxburst = 1; + slave_config->dst_maxburst = 1; + + return 0; +} + +static const struct snd_dmaengine_pcm_config +atmel_pdmic_dmaengine_pcm_config = { + .prepare_slave_config = atmel_pdmic_platform_configure_dma, + .pcm_hardware = &atmel_pdmic_hw, + .prealloc_buffer_size = ATMEL_PDMIC_PREALLOC_BUF_SIZE, +}; + +/* codec */ +/* Mic Gain = dgain * 2^(-scale) */ +struct mic_gain { + unsigned int dgain; + unsigned int scale; +}; + +/* range from -90 dB to 90 dB */ +static const struct mic_gain mic_gain_table[] = { +{ 1, 15}, { 1, 14}, /* -90, -84 dB */ +{ 3, 15}, { 1, 13}, { 3, 14}, { 1, 12}, /* -81, -78, -75, -72 dB */ +{ 5, 14}, { 13, 15}, /* -70, -68 dB */ +{ 9, 14}, { 21, 15}, { 23, 15}, { 13, 14}, /* -65 ~ -62 dB */ +{ 29, 15}, { 33, 15}, { 37, 15}, { 41, 15}, /* -61 ~ -58 dB */ +{ 23, 14}, { 13, 13}, { 58, 15}, { 65, 15}, /* -57 ~ -54 dB */ +{ 73, 15}, { 41, 14}, { 23, 13}, { 13, 12}, /* -53 ~ -50 dB */ +{ 29, 13}, { 65, 14}, { 73, 14}, { 41, 13}, /* -49 ~ -46 dB */ +{ 23, 12}, { 207, 15}, { 29, 12}, { 65, 13}, /* -45 ~ -42 dB */ +{ 73, 13}, { 41, 12}, { 23, 11}, { 413, 15}, /* -41 ~ -38 dB */ +{ 463, 15}, { 519, 15}, { 583, 15}, { 327, 14}, /* -37 ~ -34 dB */ +{ 367, 14}, { 823, 15}, { 231, 13}, { 1036, 15}, /* -33 ~ -30 dB */ +{ 1163, 15}, { 1305, 15}, { 183, 12}, { 1642, 15}, /* -29 ~ -26 dB */ +{ 1843, 15}, { 2068, 15}, { 145, 11}, { 2603, 15}, /* -25 ~ -22 dB */ +{ 365, 12}, { 3277, 15}, { 3677, 15}, { 4125, 15}, /* -21 ~ -18 dB */ +{ 4629, 15}, { 5193, 15}, { 5827, 15}, { 3269, 14}, /* -17 ~ -14 dB */ +{ 917, 12}, { 8231, 15}, { 9235, 15}, { 5181, 14}, /* -13 ~ -10 dB */ +{11627, 15}, {13045, 15}, {14637, 15}, {16423, 15}, /* -9 ~ -6 dB */ +{18427, 15}, {20675, 15}, { 5799, 13}, {26029, 15}, /* -5 ~ -2 dB */ +{ 7301, 13}, { 1, 0}, {18383, 14}, {10313, 13}, /* -1 ~ 2 dB */ +{23143, 14}, {25967, 14}, {29135, 14}, {16345, 13}, /* 3 ~ 6 dB */ +{ 4585, 11}, {20577, 13}, { 1443, 9}, {25905, 13}, /* 7 ~ 10 dB */ +{14533, 12}, { 8153, 11}, { 2287, 9}, {20529, 12}, /* 11 ~ 14 dB */ +{11517, 11}, { 6461, 10}, {28997, 12}, { 4067, 9}, /* 15 ~ 18 dB */ +{18253, 11}, { 10, 0}, {22979, 11}, {25783, 11}, /* 19 ~ 22 dB */ +{28929, 11}, {32459, 11}, { 9105, 9}, {20431, 10}, /* 23 ~ 26 dB */ +{22925, 10}, {12861, 9}, { 7215, 8}, {16191, 9}, /* 27 ~ 30 dB */ +{ 9083, 8}, {20383, 9}, {11435, 8}, { 6145, 7}, /* 31 ~ 34 dB */ +{ 3599, 6}, {32305, 9}, {18123, 8}, {20335, 8}, /* 35 ~ 38 dB */ +{ 713, 3}, { 100, 0}, { 7181, 6}, { 8057, 6}, /* 39 ~ 42 dB */ +{ 565, 2}, {20287, 7}, {11381, 6}, {25539, 7}, /* 43 ~ 46 dB */ +{ 1791, 3}, { 4019, 4}, { 9019, 5}, {20239, 6}, /* 47 ~ 50 dB */ +{ 5677, 4}, {25479, 6}, { 7147, 4}, { 8019, 4}, /* 51 ~ 54 dB */ +{17995, 5}, {20191, 5}, {11327, 4}, {12709, 4}, /* 55 ~ 58 dB */ +{ 3565, 2}, { 1000, 0}, { 1122, 0}, { 1259, 0}, /* 59 ~ 62 dB */ +{ 2825, 1}, {12679, 3}, { 7113, 2}, { 7981, 2}, /* 63 ~ 66 dB */ +{ 8955, 2}, {20095, 3}, {22547, 3}, {12649, 2}, /* 67 ~ 70 dB */ +{28385, 3}, { 3981, 0}, {17867, 2}, {20047, 2}, /* 71 ~ 74 dB */ +{11247, 1}, {12619, 1}, {14159, 1}, {31773, 2}, /* 75 ~ 78 dB */ +{17825, 1}, {10000, 0}, {11220, 0}, {12589, 0}, /* 79 ~ 82 dB */ +{28251, 1}, {15849, 0}, {17783, 0}, {19953, 0}, /* 83 ~ 86 dB */ +{22387, 0}, {25119, 0}, {28184, 0}, {31623, 0}, /* 87 ~ 90 dB */ +}; + +static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-9000, 600, 0), + 2, 5, TLV_DB_SCALE_ITEM(-8100, 300, 0), + 6, 7, TLV_DB_SCALE_ITEM(-7000, 200, 0), + 8, ARRAY_SIZE(mic_gain_table)-1, TLV_DB_SCALE_ITEM(-6500, 100, 0), +); + +int pdmic_get_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int dgain_val, scale_val; + int i; + + dgain_val = (snd_soc_read(codec, PDMIC_DSPR1) & PDMIC_DSPR1_DGAIN_MASK) + >> PDMIC_DSPR1_DGAIN_SHIFT; + + scale_val = (snd_soc_read(codec, PDMIC_DSPR0) & PDMIC_DSPR0_SCALE_MASK) + >> PDMIC_DSPR0_SCALE_SHIFT; + + for (i = 0; i < ARRAY_SIZE(mic_gain_table); i++) { + if ((mic_gain_table[i].dgain == dgain_val) && + (mic_gain_table[i].scale == scale_val)) + ucontrol->value.integer.value[0] = i; + } + + return 0; +} + +static int pdmic_put_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + int max = mc->max; + unsigned int val; + int ret; + + val = ucontrol->value.integer.value[0]; + + if (val > max) + return -EINVAL; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_MASK, + mic_gain_table[val].dgain << PDMIC_DSPR1_DGAIN_SHIFT); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR0, PDMIC_DSPR0_SCALE_MASK, + mic_gain_table[val].scale << PDMIC_DSPR0_SCALE_SHIFT); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_kcontrol_new atmel_pdmic_snd_controls[] = { +SOC_SINGLE_EXT_TLV("Mic Capture Volume", PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_SHIFT, + ARRAY_SIZE(mic_gain_table)-1, 0, + pdmic_get_mic_volsw, pdmic_put_mic_volsw, mic_gain_tlv), + +SOC_SINGLE("High Pass Filter Switch", PDMIC_DSPR0, + PDMIC_DSPR0_HPFBYP_SHIFT, 1, 1), + +SOC_SINGLE("SINCC Filter Switch", PDMIC_DSPR0, PDMIC_DSPR0_SINBYP_SHIFT, 1, 1), +}; + +static int atmel_pdmic_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_OFFSET_MASK, + (u32)(dd->pdata->mic_offset << PDMIC_DSPR1_OFFSET_SHIFT)); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pdmic = { + .probe = atmel_pdmic_codec_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), +}; + +/* codec dai component */ +#define PDMIC_MR_PRESCAL_MAX_VAL 127 + +static int +atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int rate_min = substream->runtime->hw.rate_min; + unsigned int rate_max = substream->runtime->hw.rate_max; + int fs = params_rate(params); + int bits = params_width(params); + unsigned long pclk_rate, gclk_rate; + unsigned int f_pdmic; + u32 mr_val, dspr0_val, pclk_prescal, gclk_prescal; + + if (params_channels(params) != 1) { + dev_err(codec->dev, + "only supports one channel\n"); + return -EINVAL; + } + + if ((fs < rate_min) || (fs > rate_max)) { + dev_err(codec->dev, + "sample rate is %dHz, min rate is %dHz, max rate is %dHz\n", + fs, rate_min, rate_max); + + return -EINVAL; + } + + switch (bits) { + case 16: + dspr0_val = (PDMIC_DSPR0_SIZE_16_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + case 32: + dspr0_val = (PDMIC_DSPR0_SIZE_32_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + default: + return -EINVAL; + } + + if ((fs << 7) > (rate_max << 6)) { + f_pdmic = fs << 6; + dspr0_val |= PDMIC_DSPR0_OSR_64 << PDMIC_DSPR0_OSR_SHIFT; + } else { + f_pdmic = fs << 7; + dspr0_val |= PDMIC_DSPR0_OSR_128 << PDMIC_DSPR0_OSR_SHIFT; + } + + pclk_rate = clk_get_rate(dd->pclk); + gclk_rate = clk_get_rate(dd->gclk); + + /* PRESCAL = SELCK/(2*f_pdmic) - 1*/ + pclk_prescal = (u32)(pclk_rate/(f_pdmic << 1)) - 1; + gclk_prescal = (u32)(gclk_rate/(f_pdmic << 1)) - 1; + + if ((pclk_prescal > PDMIC_MR_PRESCAL_MAX_VAL) || + (gclk_rate/((gclk_prescal + 1) << 1) < + pclk_rate/((pclk_prescal + 1) << 1))) { + mr_val = gclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_GCK << PDMIC_MR_CLKS_SHIFT; + } else { + mr_val = pclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_PCK << PDMIC_MR_CLKS_SHIFT; + } + + snd_soc_update_bits(codec, PDMIC_MR, + PDMIC_MR_PRESCAL_MASK | PDMIC_MR_CLKS_MASK, mr_val); + + snd_soc_update_bits(codec, PDMIC_DSPR0, + PDMIC_DSPR0_OSR_MASK | PDMIC_DSPR0_SIZE_MASK, dspr0_val); + + return 0; +} + +static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + return 0; +} + +static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = PDMIC_CR_ENPDM_EN << PDMIC_CR_ENPDM_SHIFT; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, val); + + return 0; +} + +static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { + .hw_params = atmel_pdmic_codec_dai_hw_params, + .prepare = atmel_pdmic_codec_dai_prepare, + .trigger = atmel_pdmic_codec_dai_trigger, +}; + +#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" + +static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { + .name = ATMEL_PDMIC_CODEC_DAI_NAME, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS, + }, + .ops = &atmel_pdmic_codec_dai_ops, +}; + +/* ASoC sound card */ +static int atmel_pdmic_asoc_card_init(struct device *dev, + struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL); + if (!dai_link) + return -ENOMEM; + + dai_link->name = "PDMIC"; + dai_link->stream_name = "PDMIC PCM"; + dai_link->codec_dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->cpu_dai_name = dev_name(dev); + dai_link->codec_name = dev_name(dev); + dai_link->platform_name = dev_name(dev); + + card->dai_link = dai_link; + card->num_links = 1; + card->name = dd->pdata->card_name; + card->dev = dev; + + return 0; +} + +static void atmel_pdmic_get_sample_rate(struct atmel_pdmic *dd, + unsigned int *rate_min, unsigned int *rate_max) +{ + u32 mic_min_freq = dd->pdata->mic_min_freq; + u32 mic_max_freq = dd->pdata->mic_max_freq; + u32 clk_max_rate = (u32)(clk_get_rate(dd->pclk) >> 1); + u32 clk_min_rate = (u32)(clk_get_rate(dd->gclk) >> 8); + + if (mic_max_freq > clk_max_rate) + mic_max_freq = clk_max_rate; + + if (mic_min_freq < clk_min_rate) + mic_min_freq = clk_min_rate; + + *rate_min = DIV_ROUND_CLOSEST(mic_min_freq, 128); + *rate_max = mic_max_freq >> 6; +} + +/* PDMIC interrupt handler */ +static irqreturn_t atmel_pdmic_interrupt(int irq, void *dev_id) +{ + struct atmel_pdmic *dd = (struct atmel_pdmic *)dev_id; + u32 pdmic_isr; + irqreturn_t ret = IRQ_NONE; + + regmap_read(dd->regmap, PDMIC_ISR, &pdmic_isr); + + if (pdmic_isr & PDMIC_ISR_OVRE) { + regmap_update_bits(dd->regmap, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + snd_pcm_stop_xrun(dd->substream); + + ret = IRQ_HANDLED; + } + + return ret; +} + +/* regmap configuration */ +#define ATMEL_PDMIC_REG_MAX 0x124 +static const struct regmap_config atmel_pdmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = ATMEL_PDMIC_REG_MAX, +}; + +static int atmel_pdmic_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct atmel_pdmic *dd; + struct resource *res; + void __iomem *io_base; + const struct atmel_pdmic_pdata *pdata; + struct snd_soc_card *card; + unsigned int rate_min, rate_max; + int ret; + + pdata = atmel_pdmic_dt_init(dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); + + dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL); + if (!dd) + return -ENOMEM; + + dd->pdata = pdata; + + dd->irq = platform_get_irq(pdev, 0); + if (dd->irq < 0) { + ret = dd->irq; + dev_err(dev, "failed to could not get irq: %d\n", ret); + return ret; + } + + dd->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(dd->pclk)) { + ret = PTR_ERR(dd->pclk); + dev_err(dev, "failed to get peripheral clock: %d\n", ret); + return ret; + } + + dd->gclk = devm_clk_get(dev, "gclk"); + if (IS_ERR(dd->gclk)) { + ret = PTR_ERR(dd->gclk); + dev_err(dev, "failed to get GCK: %d\n", ret); + return ret; + } + + /* The gclk clock frequency must always be tree times + * lower than the pclk clock frequency + */ + ret = clk_set_rate(dd->gclk, clk_get_rate(dd->pclk)/3); + if (ret) { + dev_err(dev, "failed to set GCK clock rate: %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "no memory resource\n"); + return -ENXIO; + } + + io_base = devm_ioremap_resource(dev, res); + if (IS_ERR(io_base)) { + ret = PTR_ERR(io_base); + dev_err(dev, "failed to remap register memory: %d\n", ret); + return ret; + } + + dd->phy_base = res->start; + + dd->regmap = devm_regmap_init_mmio(dev, io_base, + &atmel_pdmic_regmap_config); + if (IS_ERR(dd->regmap)) { + ret = PTR_ERR(dd->regmap); + dev_err(dev, "failed to init register map: %d\n", ret); + return ret; + } + + ret = devm_request_irq(dev, dd->irq, atmel_pdmic_interrupt, 0, + "PDMIC", (void *)dd); + if (ret < 0) { + dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", + dd->irq, ret); + return ret; + } + + /* Get the minimal and maximal sample rate that micphone supports */ + atmel_pdmic_get_sample_rate(dd, &rate_min, &rate_max); + + /* register cpu dai */ + atmel_pdmic_cpu_dai.capture.rate_min = rate_min; + atmel_pdmic_cpu_dai.capture.rate_max = rate_max; + ret = devm_snd_soc_register_component(dev, + &atmel_pdmic_cpu_dai_component, + &atmel_pdmic_cpu_dai, 1); + if (ret) { + dev_err(dev, "could not register CPU DAI: %d\n", ret); + return ret; + } + + /* register platform */ + ret = devm_snd_dmaengine_pcm_register(dev, + &atmel_pdmic_dmaengine_pcm_config, + 0); + if (ret) { + dev_err(dev, "could not register platform: %d\n", ret); + return ret; + } + + /* register codec and codec dai */ + atmel_pdmic_codec_dai.capture.rate_min = rate_min; + atmel_pdmic_codec_dai.capture.rate_max = rate_max; + ret = snd_soc_register_codec(dev, &soc_codec_dev_pdmic, + &atmel_pdmic_codec_dai, 1); + if (ret) { + dev_err(dev, "could not register codec: %d\n", ret); + return ret; + } + + /* register sound card */ + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } + + snd_soc_card_set_drvdata(card, dd); + platform_set_drvdata(pdev, card); + + ret = atmel_pdmic_asoc_card_init(dev, card); + if (ret) { + dev_err(dev, "failed to init sound card: %d\n", ret); + goto unregister_codec; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) { + dev_err(dev, "failed to register sound card: %d\n", ret); + goto unregister_codec; + } + + return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; +} + +static int atmel_pdmic_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver atmel_pdmic_driver = { + .driver = { + .name = "atmel-pdmic", + .of_match_table = of_match_ptr(atmel_pdmic_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = atmel_pdmic_probe, + .remove = atmel_pdmic_remove, +}; +module_platform_driver(atmel_pdmic_driver); + +MODULE_DESCRIPTION("Atmel PDMIC driver under ALSA SoC architecture"); +MODULE_AUTHOR("Songjun Wu "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/atmel/atmel-pdmic.h b/sound/soc/atmel/atmel-pdmic.h new file mode 100644 index 000000000000..4527ac741919 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.h @@ -0,0 +1,80 @@ +#ifndef __ATMEL_PDMIC_H_ +#define __ATMEL_PDMIC_H_ + +#include + +#define PDMIC_CR 0x00000000 + +#define PDMIC_CR_SWRST 0x1 +#define PDMIC_CR_SWRST_MASK BIT(0) +#define PDMIC_CR_SWRST_SHIFT (0) + +#define PDMIC_CR_ENPDM_DIS 0x0 +#define PDMIC_CR_ENPDM_EN 0x1 +#define PDMIC_CR_ENPDM_MASK BIT(4) +#define PDMIC_CR_ENPDM_SHIFT (4) + +#define PDMIC_MR 0x00000004 + +#define PDMIC_MR_CLKS_PCK 0x0 +#define PDMIC_MR_CLKS_GCK 0x1 +#define PDMIC_MR_CLKS_MASK BIT(4) +#define PDMIC_MR_CLKS_SHIFT (4) + +#define PDMIC_MR_PRESCAL_MASK GENMASK(14, 8) +#define PDMIC_MR_PRESCAL_SHIFT (8) + +#define PDMIC_CDR 0x00000014 + +#define PDMIC_IER 0x00000018 +#define PDMIC_IER_OVRE BIT(25) + +#define PDMIC_IDR 0x0000001c +#define PDMIC_IDR_OVRE BIT(25) + +#define PDMIC_IMR 0x00000020 + +#define PDMIC_ISR 0x00000024 +#define PDMIC_ISR_OVRE BIT(25) + +#define PDMIC_DSPR0 0x00000058 + +#define PDMIC_DSPR0_HPFBYP_DIS 0x1 +#define PDMIC_DSPR0_HPFBYP_EN 0x0 +#define PDMIC_DSPR0_HPFBYP_MASK BIT(1) +#define PDMIC_DSPR0_HPFBYP_SHIFT (1) + +#define PDMIC_DSPR0_SINBYP_DIS 0x1 +#define PDMIC_DSPR0_SINBYP_EN 0x0 +#define PDMIC_DSPR0_SINBYP_MASK BIT(2) +#define PDMIC_DSPR0_SINBYP_SHIFT (2) + +#define PDMIC_DSPR0_SIZE_16_BITS 0x0 +#define PDMIC_DSPR0_SIZE_32_BITS 0x1 +#define PDMIC_DSPR0_SIZE_MASK BIT(3) +#define PDMIC_DSPR0_SIZE_SHIFT (3) + +#define PDMIC_DSPR0_OSR_128 0x0 +#define PDMIC_DSPR0_OSR_64 0x1 +#define PDMIC_DSPR0_OSR_MASK GENMASK(6, 4) +#define PDMIC_DSPR0_OSR_SHIFT (4) + +#define PDMIC_DSPR0_SCALE_MASK GENMASK(11, 8) +#define PDMIC_DSPR0_SCALE_SHIFT (8) + +#define PDMIC_DSPR0_SHIFT_MASK GENMASK(15, 12) +#define PDMIC_DSPR0_SHIFT_SHIFT (12) + +#define PDMIC_DSPR1 0x0000005c + +#define PDMIC_DSPR1_DGAIN_MASK GENMASK(14, 0) +#define PDMIC_DSPR1_DGAIN_SHIFT (0) + +#define PDMIC_DSPR1_OFFSET_MASK GENMASK(31, 16) +#define PDMIC_DSPR1_OFFSET_SHIFT (16) + +#define PDMIC_WPMR 0x000000e4 + +#define PDMIC_WPSR 0x000000e8 + +#endif From 4bc5145cec89b3df719d87d0a6876aa00941d66c Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Thu, 17 Dec 2015 17:50:00 +0800 Subject: [PATCH 15/23] ASoC: atmel-classd: DT binding for PDMIC driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/atmel-pdmic.txt | 55 +++++++++++++++++++ 1 file changed, 55 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt diff --git a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt new file mode 100644 index 000000000000..e0875f17c229 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt @@ -0,0 +1,55 @@ +* Atmel PDMIC driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-pdmic". +- reg + Should contain PDMIC registers location and length. +- interrupts + Should contain the IRQ line for the PDMIC. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "rx". +- clock-names + Required elements: + - "pclk" peripheral clock + - "gclk" generated clock +- clocks + Must contain an entry for each required entry in clock-names. + Please refer to clock-bindings.txt. +- atmel,mic-min-freq + The minimal frequency that the micphone supports. +- atmel,mic-max-freq + The maximal frequency that the micphone supports. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound card. + The default value is "PDMIC". +- atmel,mic-offset + The offset that should be added. + The range is from -32768 to 32767. + The default value is 0. + +Example: + pdmic@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC @ sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; From 50860e1d17d1bc2f1a2ebfc5042f2af786e53ad6 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Tue, 22 Dec 2015 14:06:42 +0800 Subject: [PATCH 16/23] ASoC: atmel_wm8904: add snd_soc_pm_ops Sometimes the audio play can not be resumed after it is suspended. Add snd_soc_pm_ops to execute power management operations, then this issue is fixed. Signed-off-by: Songjun Wu Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 1933bcd46cca..fdd28ed3e0b9 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -183,6 +183,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .driver = { .name = "atmel-wm8904-audio", .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, .remove = atmel_asoc_wm8904_remove, From 1d981e0a5af78339d55085041c6eb3b9a8626920 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:42 +0000 Subject: [PATCH 17/23] ASoC: wm5110: Provide basic hookup for voice control Register a platform driver for the CODEC and add DAIs that will be used to connect a compressed record path for the voice control functionality. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 46 +++++++++++++++++++++++++++++++++++++- 3 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 55e14a3ed5e1..4f482f7b63fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -214,6 +214,7 @@ config SND_SOC_WM_HUBS config SND_SOC_WM_ADSP tristate + select SND_SOC_COMPRESS default y if SND_SOC_CS47L24=y default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b4f1867ae9d6..8b6adb5419bb 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -57,7 +57,7 @@ #define ARIZONA_CLK_98MHZ 5 #define ARIZONA_CLK_147MHZ 6 -#define ARIZONA_MAX_DAI 6 +#define ARIZONA_MAX_DAI 8 #define ARIZONA_MAX_ADSP 4 #define ARIZONA_DVFS_SR1_RQ 0x001 diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e93e5420943e..67d56511699a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1810,6 +1810,9 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "Slim2 Capture", NULL, "SYSCLK" }, { "Slim3 Capture", NULL, "SYSCLK" }, + { "Voice Control DSP", NULL, "DSP3" }, + { "Voice Control DSP", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -2132,6 +2135,27 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_simple_dai_ops, }, + { + .name = "wm5110-cpu-voicectrl", + .capture = { + .stream_name = "Voice Control CPU", + .channels_min = 1, + .channels_max = 1, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "wm5110-dsp-voicectrl", + .capture = { + .stream_name = "Voice Control DSP", + .channels_min = 1, + .channels_max = 1, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + }, }; static int wm5110_codec_probe(struct snd_soc_codec *codec) @@ -2224,6 +2248,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes), }; +static struct snd_compr_ops wm5110_compr_ops = { +}; + +static struct snd_soc_platform_driver wm5110_compr_platform = { + .compr_ops = &wm5110_compr_ops, +}; + static int wm5110_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); @@ -2284,8 +2315,21 @@ static int wm5110_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, + ret = snd_soc_register_platform(&pdev->dev, &wm5110_compr_platform); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); + goto error; + } + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, wm5110_dai, ARRAY_SIZE(wm5110_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + snd_soc_unregister_platform(&pdev->dev); + } + +error: + return ret; } static int wm5110_remove(struct platform_device *pdev) From 14197095e14a4ad2afb6c8c1ca8e41852382481d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:43 +0000 Subject: [PATCH 18/23] ASoC: wm_adsp: Factor out finding the location of an algorithm region Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 35 +++++++++++++++++++++-------------- 1 file changed, 21 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d1e0826c7db2..27abad9c6e73 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1365,6 +1365,19 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, return alg; } +static struct wm_adsp_alg_region * + wm_adsp_find_alg_region(struct wm_adsp *dsp, int type, unsigned int id) +{ + struct wm_adsp_alg_region *alg_region; + + list_for_each_entry(alg_region, &dsp->alg_regions, list) { + if (id == alg_region->alg && type == alg_region->type) + return alg_region; + } + + return NULL; +} + static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp, int type, __be32 id, __be32 base) @@ -1737,22 +1750,16 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) break; } - reg = 0; - list_for_each_entry(alg_region, - &dsp->alg_regions, list) { - if (le32_to_cpu(blk->id) == alg_region->alg && - type == alg_region->type) { - reg = alg_region->base; - reg = wm_adsp_region_to_reg(mem, - reg); - reg += offset; - break; - } - } - - if (reg == 0) + alg_region = wm_adsp_find_alg_region(dsp, type, + le32_to_cpu(blk->id)); + if (alg_region) { + reg = alg_region->base; + reg = wm_adsp_region_to_reg(mem, reg); + reg += offset; + } else { adsp_err(dsp, "No %x for algorithm %x\n", type, le32_to_cpu(blk->id)); + } break; default: From dbb6b94339e82ad2532798ed80f2651d21d97975 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:44 +0000 Subject: [PATCH 19/23] ALSA: compress: Add SND_AUDIOCODEC_BESPOKE When working with the compressed framework occasionally vendors will use esoteric internal audio formats. For such formats it doesn't really make sense to add an new define to the kernel as their use is not sufficiently general. This patch adds a new define SND_AUDIOCODEC_BESPOKE that vendors can use in such situations. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/compress_params.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index d9bd9ca0d5b0..9625484a4a2a 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -73,7 +73,8 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE /* * Profile and modes are listed with bit masks. This allows for a @@ -312,7 +313,7 @@ struct snd_enc_flac { struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ - __s32 reserved[15]; + __s32 reserved[15]; /* Can be used for SND_AUDIOCODEC_BESPOKE */ } __attribute__((packed, aligned(4))); union snd_codec_options { From 406abc95a0397e10eb6edcfe824b1a8bf6578a0b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:45 +0000 Subject: [PATCH 20/23] ASoC: wm_adsp: Add support for opening a compressed stream Allow user-space to open a compressed stream, although no data will be passed yet, as part of this adding the ability to define supported capabilities per firmware and check these match the stream being opened. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 23 +++++ sound/soc/codecs/wm_adsp.c | 194 ++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +++ 3 files changed, 227 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 67d56511699a..8c0fd9106be0 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2158,6 +2158,25 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, }; +static int wm5110_open(struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct wm5110_priv *priv = snd_soc_codec_get_drvdata(rtd->codec); + struct arizona *arizona = priv->core.arizona; + int n_adsp; + + if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) { + n_adsp = 2; + } else { + dev_err(arizona->dev, + "No suitable compressed stream for DAI '%s'\n", + rtd->codec_dai->name); + return -EINVAL; + } + + return wm_adsp_compr_open(&priv->core.adsp[n_adsp], stream); +} + static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); @@ -2249,6 +2268,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { }; static struct snd_compr_ops wm5110_compr_ops = { + .open = wm5110_open, + .free = wm_adsp_compr_free, + .set_params = wm_adsp_compr_set_params, + .get_caps = wm_adsp_compr_get_caps, }; static struct snd_soc_platform_driver wm5110_compr_platform = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 27abad9c6e73..d81ed218918e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -229,8 +229,42 @@ static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { [WM_ADSP_FW_MISC] = "Misc", }; -static struct { +struct wm_adsp_compr { + struct wm_adsp *dsp; + + struct snd_compr_stream *stream; + struct snd_compressed_buffer size; +}; + +#define WM_ADSP_DATA_WORD_SIZE 3 + +#define WM_ADSP_MIN_FRAGMENTS 1 +#define WM_ADSP_MAX_FRAGMENTS 256 +#define WM_ADSP_MIN_FRAGMENT_SIZE (64 * WM_ADSP_DATA_WORD_SIZE) +#define WM_ADSP_MAX_FRAGMENT_SIZE (4096 * WM_ADSP_DATA_WORD_SIZE) + +struct wm_adsp_fw_caps { + u32 id; + struct snd_codec_desc desc; +}; + +static const struct wm_adsp_fw_caps ez2control_caps[] = { + { + .id = SND_AUDIOCODEC_BESPOKE, + .desc = { + .max_ch = 1, + .sample_rates = { 16000 }, + .num_sample_rates = 1, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static const struct { const char *file; + int compr_direction; + int num_caps; + const struct wm_adsp_fw_caps *caps; } wm_adsp_fw[WM_ADSP_NUM_FW] = { [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, [WM_ADSP_FW_HIFI] = { .file = "hifi" }, @@ -238,7 +272,12 @@ static struct { [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, [WM_ADSP_FW_RX] = { .file = "rx" }, [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, - [WM_ADSP_FW_CTRL] = { .file = "ctrl" }, + [WM_ADSP_FW_CTRL] = { + .file = "ctrl", + .compr_direction = SND_COMPRESS_CAPTURE, + .num_caps = ARRAY_SIZE(ez2control_caps), + .caps = ez2control_caps, + }, [WM_ADSP_FW_ASR] = { .file = "asr" }, [WM_ADSP_FW_TRACE] = { .file = "trace" }, [WM_ADSP_FW_SPK_PROT] = { .file = "spk-prot" }, @@ -461,7 +500,7 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, mutex_lock(&dsp[e->shift_l].pwr_lock); - if (dsp[e->shift_l].running) + if (dsp[e->shift_l].running || dsp[e->shift_l].compr) ret = -EBUSY; else dsp[e->shift_l].fw = ucontrol->value.integer.value[0]; @@ -2178,4 +2217,153 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) +{ + struct wm_adsp_compr *compr; + int ret = 0; + + mutex_lock(&dsp->pwr_lock); + + if (wm_adsp_fw[dsp->fw].num_caps == 0) { + adsp_err(dsp, "Firmware does not support compressed API\n"); + ret = -ENXIO; + goto out; + } + + if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) { + adsp_err(dsp, "Firmware does not support stream direction\n"); + ret = -EINVAL; + goto out; + } + + compr = kzalloc(sizeof(*compr), GFP_KERNEL); + if (!compr) { + ret = -ENOMEM; + goto out; + } + + compr->dsp = dsp; + compr->stream = stream; + + dsp->compr = compr; + + stream->runtime->private_data = compr; + +out: + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_open); + +int wm_adsp_compr_free(struct snd_compr_stream *stream) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + struct wm_adsp *dsp = compr->dsp; + + mutex_lock(&dsp->pwr_lock); + + dsp->compr = NULL; + + kfree(compr); + + mutex_unlock(&dsp->pwr_lock); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_free); + +static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + struct wm_adsp *dsp = compr->dsp; + const struct wm_adsp_fw_caps *caps; + const struct snd_codec_desc *desc; + int i, j; + + if (params->buffer.fragment_size < WM_ADSP_MIN_FRAGMENT_SIZE || + params->buffer.fragment_size > WM_ADSP_MAX_FRAGMENT_SIZE || + params->buffer.fragments < WM_ADSP_MIN_FRAGMENTS || + params->buffer.fragments > WM_ADSP_MAX_FRAGMENTS || + params->buffer.fragment_size % WM_ADSP_DATA_WORD_SIZE) { + adsp_err(dsp, "Invalid buffer fragsize=%d fragments=%d\n", + params->buffer.fragment_size, + params->buffer.fragments); + + return -EINVAL; + } + + for (i = 0; i < wm_adsp_fw[dsp->fw].num_caps; i++) { + caps = &wm_adsp_fw[dsp->fw].caps[i]; + desc = &caps->desc; + + if (caps->id != params->codec.id) + continue; + + if (stream->direction == SND_COMPRESS_PLAYBACK) { + if (desc->max_ch < params->codec.ch_out) + continue; + } else { + if (desc->max_ch < params->codec.ch_in) + continue; + } + + if (!(desc->formats & (1 << params->codec.format))) + continue; + + for (j = 0; j < desc->num_sample_rates; ++j) + if (desc->sample_rates[j] == params->codec.sample_rate) + return 0; + } + + adsp_err(dsp, "Invalid params id=%u ch=%u,%u rate=%u fmt=%u\n", + params->codec.id, params->codec.ch_in, params->codec.ch_out, + params->codec.sample_rate, params->codec.format); + return -EINVAL; +} + +int wm_adsp_compr_set_params(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + int ret; + + ret = wm_adsp_compr_check_params(stream, params); + if (ret) + return ret; + + compr->size = params->buffer; + + adsp_dbg(compr->dsp, "fragment_size=%d fragments=%d\n", + compr->size.fragment_size, compr->size.fragments); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_set_params); + +int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, + struct snd_compr_caps *caps) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + int fw = compr->dsp->fw; + int i; + + if (wm_adsp_fw[fw].caps) { + for (i = 0; i < wm_adsp_fw[fw].num_caps; i++) + caps->codecs[i] = wm_adsp_fw[fw].caps[i].id; + + caps->num_codecs = i; + caps->direction = wm_adsp_fw[fw].compr_direction; + + caps->min_fragment_size = WM_ADSP_MIN_FRAGMENT_SIZE; + caps->max_fragment_size = WM_ADSP_MAX_FRAGMENT_SIZE; + caps->min_fragments = WM_ADSP_MIN_FRAGMENTS; + caps->max_fragments = WM_ADSP_MAX_FRAGMENTS; + } + + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_get_caps); + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d2a8c78ed50b..33c9b5283d26 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -15,6 +15,7 @@ #include #include +#include #include "wmfw.h" @@ -30,6 +31,8 @@ struct wm_adsp_alg_region { unsigned int base; }; +struct wm_adsp_compr; + struct wm_adsp { const char *part; int num; @@ -59,6 +62,8 @@ struct wm_adsp { struct work_struct boot_work; + struct wm_adsp_compr *compr; + struct mutex pwr_lock; #ifdef CONFIG_DEBUG_FS @@ -97,4 +102,12 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int wm_adsp_compr_open(struct wm_adsp *dsp, + struct snd_compr_stream *stream); +extern int wm_adsp_compr_free(struct snd_compr_stream *stream); +extern int wm_adsp_compr_set_params(struct snd_compr_stream *stream, + struct snd_compr_params *params); +extern int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, + struct snd_compr_caps *caps); + #endif From 2cd19bdbf83c4c70b2ee36d022c5ded2738d2e19 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:46 +0000 Subject: [PATCH 21/23] ASoC: wm_adsp: Add code to locate and initialise compressed buffer Add code that locates and initialises the buffer of compressed data on the DSP if the firmware supported compressed data capture. The buffer struct (wm_adsp_compr_buf) is kept separate from the stream struct (wm_adsp_compr) this will allow much easier support of multiple streams of data from the one DSP in the future, although support for this will not be added in this patch chain. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 291 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 2 + 2 files changed, 293 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d81ed218918e..90994a5528a4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -229,6 +229,58 @@ static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { [WM_ADSP_FW_MISC] = "Misc", }; +struct wm_adsp_system_config_xm_hdr { + __be32 sys_enable; + __be32 fw_id; + __be32 fw_rev; + __be32 boot_status; + __be32 watchdog; + __be32 dma_buffer_size; + __be32 rdma[6]; + __be32 wdma[8]; + __be32 build_job_name[3]; + __be32 build_job_number; +}; + +struct wm_adsp_alg_xm_struct { + __be32 magic; + __be32 smoothing; + __be32 threshold; + __be32 host_buf_ptr; + __be32 start_seq; + __be32 high_water_mark; + __be32 low_water_mark; + __be64 smoothed_power; +}; + +struct wm_adsp_buffer { + __be32 X_buf_base; /* XM base addr of first X area */ + __be32 X_buf_size; /* Size of 1st X area in words */ + __be32 X_buf_base2; /* XM base addr of 2nd X area */ + __be32 X_buf_brk; /* Total X size in words */ + __be32 Y_buf_base; /* YM base addr of Y area */ + __be32 wrap; /* Total size X and Y in words */ + __be32 high_water_mark; /* Point at which IRQ is asserted */ + __be32 irq_count; /* bits 1-31 count IRQ assertions */ + __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ + __be32 next_write_index; /* word index of next write */ + __be32 next_read_index; /* word index of next read */ + __be32 error; /* error if any */ + __be32 oldest_block_index; /* word index of oldest surviving */ + __be32 requested_rewind; /* how many blocks rewind was done */ + __be32 reserved_space; /* internal */ + __be32 min_free; /* min free space since stream start */ + __be32 blocks_written[2]; /* total blocks written (64 bit) */ + __be32 words_written[2]; /* total words written (64 bit) */ +}; + +struct wm_adsp_compr_buf { + struct wm_adsp *dsp; + + struct wm_adsp_buffer_region *regions; + u32 host_buf_ptr; +}; + struct wm_adsp_compr { struct wm_adsp *dsp; @@ -243,9 +295,53 @@ struct wm_adsp_compr { #define WM_ADSP_MIN_FRAGMENT_SIZE (64 * WM_ADSP_DATA_WORD_SIZE) #define WM_ADSP_MAX_FRAGMENT_SIZE (4096 * WM_ADSP_DATA_WORD_SIZE) +#define WM_ADSP_ALG_XM_STRUCT_MAGIC 0x49aec7 + +#define HOST_BUFFER_FIELD(field) \ + (offsetof(struct wm_adsp_buffer, field) / sizeof(__be32)) + +#define ALG_XM_FIELD(field) \ + (offsetof(struct wm_adsp_alg_xm_struct, field) / sizeof(__be32)) + +static int wm_adsp_buffer_init(struct wm_adsp *dsp); +static int wm_adsp_buffer_free(struct wm_adsp *dsp); + +struct wm_adsp_buffer_region { + unsigned int offset; + unsigned int cumulative_size; + unsigned int mem_type; + unsigned int base_addr; +}; + +struct wm_adsp_buffer_region_def { + unsigned int mem_type; + unsigned int base_offset; + unsigned int size_offset; +}; + +static struct wm_adsp_buffer_region_def ez2control_regions[] = { + { + .mem_type = WMFW_ADSP2_XM, + .base_offset = HOST_BUFFER_FIELD(X_buf_base), + .size_offset = HOST_BUFFER_FIELD(X_buf_size), + }, + { + .mem_type = WMFW_ADSP2_XM, + .base_offset = HOST_BUFFER_FIELD(X_buf_base2), + .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + }, + { + .mem_type = WMFW_ADSP2_YM, + .base_offset = HOST_BUFFER_FIELD(Y_buf_base), + .size_offset = HOST_BUFFER_FIELD(wrap), + }, +}; + struct wm_adsp_fw_caps { u32 id; struct snd_codec_desc desc; + int num_regions; + struct wm_adsp_buffer_region_def *region_defs; }; static const struct wm_adsp_fw_caps ez2control_caps[] = { @@ -257,6 +353,8 @@ static const struct wm_adsp_fw_caps ez2control_caps[] = { .num_sample_rates = 1, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .num_regions = ARRAY_SIZE(ez2control_regions), + .region_defs = ez2control_regions, }, }; @@ -2123,6 +2221,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; + + if (wm_adsp_fw[dsp->fw].num_caps != 0) + ret = wm_adsp_buffer_init(dsp); + break; case SND_SOC_DAPM_PRE_PMD: @@ -2157,6 +2259,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, kfree(alg_region); } + if (wm_adsp_fw[dsp->fw].num_caps != 0) + wm_adsp_buffer_free(dsp); + mutex_unlock(&dsp->pwr_lock); adsp_dbg(dsp, "Shutdown complete\n"); @@ -2366,4 +2471,190 @@ int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, } EXPORT_SYMBOL_GPL(wm_adsp_compr_get_caps); +static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, + unsigned int mem_addr, + unsigned int num_words, u32 *data) +{ + struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); + unsigned int i, reg; + int ret; + + if (!mem) + return -EINVAL; + + reg = wm_adsp_region_to_reg(mem, mem_addr); + + ret = regmap_raw_read(dsp->regmap, reg, data, + sizeof(*data) * num_words); + if (ret < 0) + return ret; + + for (i = 0; i < num_words; ++i) + data[i] = be32_to_cpu(data[i]) & 0x00ffffffu; + + return 0; +} + +static inline int wm_adsp_read_data_word(struct wm_adsp *dsp, int mem_type, + unsigned int mem_addr, u32 *data) +{ + return wm_adsp_read_data_block(dsp, mem_type, mem_addr, 1, data); +} + +static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, + unsigned int mem_addr, u32 data) +{ + struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); + unsigned int reg; + + if (!mem) + return -EINVAL; + + reg = wm_adsp_region_to_reg(mem, mem_addr); + + data = cpu_to_be32(data & 0x00ffffffu); + + return regmap_raw_write(dsp->regmap, reg, &data, sizeof(data)); +} + +static inline int wm_adsp_buffer_read(struct wm_adsp_compr_buf *buf, + unsigned int field_offset, u32 *data) +{ + return wm_adsp_read_data_word(buf->dsp, WMFW_ADSP2_XM, + buf->host_buf_ptr + field_offset, data); +} + +static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, + unsigned int field_offset, u32 data) +{ + return wm_adsp_write_data_word(buf->dsp, WMFW_ADSP2_XM, + buf->host_buf_ptr + field_offset, data); +} + +static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp_alg_region *alg_region; + struct wm_adsp *dsp = buf->dsp; + u32 xmalg, addr, magic; + int i, ret; + + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + xmalg = sizeof(struct wm_adsp_system_config_xm_hdr) / sizeof(__be32); + + addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); + ret = wm_adsp_read_data_word(dsp, WMFW_ADSP2_XM, addr, &magic); + if (ret < 0) + return ret; + + if (magic != WM_ADSP_ALG_XM_STRUCT_MAGIC) + return -EINVAL; + + addr = alg_region->base + xmalg + ALG_XM_FIELD(host_buf_ptr); + for (i = 0; i < 5; ++i) { + ret = wm_adsp_read_data_word(dsp, WMFW_ADSP2_XM, addr, + &buf->host_buf_ptr); + if (ret < 0) + return ret; + + if (buf->host_buf_ptr) + break; + + usleep_range(1000, 2000); + } + + if (!buf->host_buf_ptr) + return -EIO; + + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + + return 0; +} + +static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) +{ + const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; + struct wm_adsp_buffer_region *region; + u32 offset = 0; + int i, ret; + + for (i = 0; i < caps->num_regions; ++i) { + region = &buf->regions[i]; + + region->offset = offset; + region->mem_type = caps->region_defs[i].mem_type; + + ret = wm_adsp_buffer_read(buf, caps->region_defs[i].base_offset, + ®ion->base_addr); + if (ret < 0) + return ret; + + ret = wm_adsp_buffer_read(buf, caps->region_defs[i].size_offset, + &offset); + if (ret < 0) + return ret; + + region->cumulative_size = offset; + + adsp_dbg(buf->dsp, + "region=%d type=%d base=%04x off=%04x size=%04x\n", + i, region->mem_type, region->base_addr, + region->offset, region->cumulative_size); + } + + return 0; +} + +static int wm_adsp_buffer_init(struct wm_adsp *dsp) +{ + struct wm_adsp_compr_buf *buf; + int ret; + + buf = kzalloc(sizeof(*buf), GFP_KERNEL); + if (!buf) + return -ENOMEM; + + buf->dsp = dsp; + + ret = wm_adsp_buffer_locate(buf); + if (ret < 0) { + adsp_err(dsp, "Failed to acquire host buffer: %d\n", ret); + goto err_buffer; + } + + buf->regions = kcalloc(wm_adsp_fw[dsp->fw].caps->num_regions, + sizeof(*buf->regions), GFP_KERNEL); + if (!buf->regions) { + ret = -ENOMEM; + goto err_buffer; + } + + ret = wm_adsp_buffer_populate(buf); + if (ret < 0) { + adsp_err(dsp, "Failed to populate host buffer: %d\n", ret); + goto err_regions; + } + + dsp->buffer = buf; + + return 0; + +err_regions: + kfree(buf->regions); +err_buffer: + kfree(buf); + return ret; +} + +static int wm_adsp_buffer_free(struct wm_adsp *dsp) +{ + if (dsp->buffer) { + kfree(dsp->buffer->regions); + kfree(dsp->buffer); + + dsp->buffer = NULL; + } + + return 0; +} + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 33c9b5283d26..0b2205a5c42f 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -32,6 +32,7 @@ struct wm_adsp_alg_region { }; struct wm_adsp_compr; +struct wm_adsp_compr_buf; struct wm_adsp { const char *part; @@ -63,6 +64,7 @@ struct wm_adsp { struct work_struct boot_work; struct wm_adsp_compr *compr; + struct wm_adsp_compr_buf *buffer; struct mutex pwr_lock; From 95fe9597d2494e8c4c9064fca1e12d1c03733ae7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 15 Dec 2015 11:29:47 +0000 Subject: [PATCH 22/23] ASoC: wm_adsp: Attach buffers and streams together The stream is created whilst the compressed stream is opened and a buffer is created when the DSP powers up. It is necessary at a point once both the DSP has powered up and the the stream has been opened to connect a stream to a buffer on the DSP. This is done in the trigger callback as this is after the DSP has been powered and obviously the stream must be open. Note that whilst the connect is currently trivial it is expected that this will get more complex when support for multiple buffers/streams per DSP is added. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm_adsp.c | 62 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 3 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8c0fd9106be0..c36409601835 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2272,6 +2272,7 @@ static struct snd_compr_ops wm5110_compr_ops = { .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, .get_caps = wm_adsp_compr_get_caps, + .trigger = wm_adsp_compr_trigger, }; static struct snd_soc_platform_driver wm5110_compr_platform = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 90994a5528a4..ac879d16c6a6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -283,6 +283,7 @@ struct wm_adsp_compr_buf { struct wm_adsp_compr { struct wm_adsp *dsp; + struct wm_adsp_compr_buf *buf; struct snd_compr_stream *stream; struct snd_compressed_buffer size; @@ -2341,6 +2342,13 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) goto out; } + if (dsp->compr) { + /* It is expect this limitation will be removed in future */ + adsp_err(dsp, "Only a single stream supported per DSP\n"); + ret = -EBUSY; + goto out; + } + compr = kzalloc(sizeof(*compr), GFP_KERNEL); if (!compr) { ret = -ENOMEM; @@ -2657,4 +2665,58 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + + return 0; +} + +int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + struct wm_adsp *dsp = compr->dsp; + int ret = 0; + + adsp_dbg(dsp, "Trigger: %d\n", cmd); + + mutex_lock(&dsp->pwr_lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (wm_adsp_compr_attached(compr)) + break; + + ret = wm_adsp_compr_attach(compr); + if (ret < 0) { + adsp_err(dsp, "Failed to link buffer and stream: %d\n", + ret); + break; + } + break; + case SNDRV_PCM_TRIGGER_STOP: + break; + default: + ret = -EINVAL; + break; + } + + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_trigger); + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0b2205a5c42f..43af093fafcf 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -111,5 +111,6 @@ extern int wm_adsp_compr_set_params(struct snd_compr_stream *stream, struct snd_compr_params *params); extern int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, struct snd_compr_caps *caps); +extern int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd); #endif From 34015f5e56c71bbdcf7189430ffb63ea67656a35 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Tue, 22 Dec 2015 15:51:39 +0100 Subject: [PATCH 23/23] ASoC: ac97: Be sure to clamp return value As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Signed-off-by: Linus Walleij Signed-off-by: Mark Brown --- sound/soc/soc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index ae563e379a72..733f5128eeff 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -92,7 +92,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) dev_dbg(codec->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); - return ret < 0 ? ret : ret & (1 << offset); + return ret < 0 ? ret : !!(ret & (1 << offset)); } static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset,