August 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RESEND v11 0/3] Add ES8328 audio codec
by Sean Cross 16 Aug '14

16 Aug '14

This patchset adds support for the Everest Semi ES8328 audio codec, used in the Novena open source laptop. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v10: - Minor documentation reformatting - Minor spacing reformatting in sound/soc/fsl/imx-audio-es8328.c Changes since v9: - Minor formatting corrections - Rework DAPM registers in es8328.h. DAPM widgets use offsets, and the v9 patch turned snd_soc_write() calls into widgets without converting from i2c masks to offsets first. - Rename DAC and ADC PLL to DLL, to reflect latest documentation. Changes since v8: - For the mclk ratios, move from an array to a lookup table, and rename the variable to be clearer. - Move all CHIPPOWER accesses into DAPM supplies. - Fix sloppy compile problems introduced in v8. - Shut down clocks and supplies on suspend, and enable them on resume. - Remove double set of ES8328_MASTERMODE_MSC. Changes since v7: - We now fetch the regulators in the device-level probe rather than in the ASoC-level probe. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 756 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 232 +++++++ 12 files changed, 1545 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH] ASoC: rt5670: add clock source selection controls
by bardliao＠realtek.com 16 Aug '14

16 Aug '14

From: Bard Liao <bardliao(a)realtek.com> We can select the clock source of some digital widgets in rt5670. This patch adds the controls of those selections. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 117 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 879d42e..91987e1 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -430,6 +430,105 @@ static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); +static const char * const rt5670_asrc_clk_source[] = { + "clk_sysy_div_out", "clk_i2s1_track", "clk_i2s2_track", + "clk_i2s3_track", "clk_i2s4_track", "clk_sys2", "clk_sys3", + "clk_sys4", "clk_sys5" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_sto_asrc_enum, RT5670_ASRC_2, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_monol_asrc_enum, RT5670_ASRC_2, + 8, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_monor_asrc_enum, RT5670_ASRC_2, + 4, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_sto1_asrc_enum, RT5670_ASRC_2, + 0, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_up_filter_asrc_enum, RT5670_ASRC_3, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_down_filter_asrc_enum, RT5670_ASRC_3, + 8, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_monol_asrc_enum, RT5670_ASRC_3, + 4, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_monor_asrc_enum, RT5670_ASRC_3, + 0, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_sto2_asrc_enum, RT5670_ASRC_5, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_asrc_enum, RT5670_DSP_CLK, + 0, rt5670_asrc_clk_source); + +static int rt5670_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5670_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5670_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5670_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5670_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5670_PWR_DAC_S1F; + break; + } + break; + case RT5670_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5670_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5670_PWR_ADC_MF_L; + break; + } + break; + case RT5670_ASRC_5: + u_bit = 0x4; + p_bit = RT5670_PWR_ADC_S2F; + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5670_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5670_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5670_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5670_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, @@ -482,6 +581,24 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), + + SOC_ENUM_EXT("DA STO Clk Sel", rt5670_da_sto_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("DA MONOL Clk Sel", rt5670_da_monol_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("DA MONOR Clk Sel", rt5670_da_monor_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD STO1 Clk Sel", rt5670_ad_sto1_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD MONOL Clk Sel", rt5670_ad_monol_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD MONOR Clk Sel", rt5670_ad_monor_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM("UP Clk Sel", rt5670_up_filter_asrc_enum), + SOC_ENUM("DOWN Clk Sel", rt5670_down_filter_asrc_enum), + SOC_ENUM_EXT("AD STO2 Clk Sel", rt5670_ad_sto2_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM("DSP Clk Sel", rt5670_dsp_asrc_enum), }; /** -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/3] ASoC: rt5677: Remove the redundant definition in head file
by Oder Chiou 16 Aug '14

16 Aug '14

The patch removes the redundant definition in head file Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.h | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 08252e7..863393e 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1393,13 +1393,6 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 -/* Debug String Length */ -#define RT5677_REG_DISP_LEN 23 - -#define RT5677_NO_JACK BIT(0) -#define RT5677_HEADSET_DET BIT(1) -#define RT5677_HEADPHO_DET BIT(2) - /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH/RFC] ASoC: Drop const from struct snd_soc_dai_link *of_node members
by Sylwester Nawrocki 16 Aug '14

16 Aug '14

Dropping the const qualifiers prevents "passing argument 1 of ‘of_node_put’ discards ‘const’ qualifier from pointer target type" type warnings when compiling the code dropping reference to cpu_of_node, codec_of_node or platform_of_node with with an of_node_put() function call. This lets us to avoid casting to struct device_node * or caching variables internally in drivers just to be able to properly drop a reference to the OF node on clean up paths. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- include/sound/soc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..a8d4b79 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -881,7 +881,7 @@ struct snd_soc_dai_link { * only for codec to codec links, or systems using device tree. */ const char *cpu_name; - const struct device_node *cpu_of_node; + struct device_node *cpu_of_node; /* * You MAY specify the DAI name of the CPU DAI. If this information is * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node @@ -893,7 +893,7 @@ struct snd_soc_dai_link { * DT/OF node, but not both. */ const char *codec_name; - const struct device_node *codec_of_node; + struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; /* @@ -902,7 +902,7 @@ struct snd_soc_dai_link { * do not need a platform. */ const char *platform_name; - const struct device_node *platform_of_node; + struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2 v2] ASoC: Add HA (HEAD acoustics) DSP codec driver template
by Stefan Roese 16 Aug '14

16 Aug '14

This codec driver template represents an I2C controlled multichannel audio codec that has many typical ASoC codec driver features like volume controls, mixer stages, mux selection, output power control, in-codec audio routings, codec bias management and DAI link configuration. This driver is based on an early version provided by Jarkko Nikula. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Stefan Roese <sr(a)denx.de> Cc: Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> --- v2: - Added/changed copyright line - Changed authorship (as suggested by Jarkko the original author) - Added Thorsten as maintainer - Remove ha_dsp_hw_params() and ha_dsp_set_dai_fmt() as its not used (only needed for CODEC as clock master which is currently not suported). - Removed some unneeded include files - "const char *const foo" used instead of "const char *foo" - SOC_MIXER_ARRAY() helper macro used - Removed ha_dsp_set_bias_level() and use default implementation - Use codec->dev instead of codec->dev->parent in dev_get_regmap() - Added CODEC reset to probe - Remove "ret" in ha_dsp_i2c_probe() sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ha-dsp.c | 333 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ha-dsp.h | 50 +++++++ 4 files changed, 389 insertions(+) create mode 100644 sound/soc/codecs/ha-dsp.c create mode 100644 sound/soc/codecs/ha-dsp.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..f357988 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,6 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_BT_SCO + select SND_SOC_HA_DSP if I2C select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -343,6 +344,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HA_DSP + tristate + config SND_SOC_HDMI_CODEC tristate "HDMI stub CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..f296bec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o +snd-soc-ha-dsp-objs := ha-dsp.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -190,6 +191,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_HA_DSP) += snd-soc-ha-dsp.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ha-dsp.c b/sound/soc/codecs/ha-dsp.c new file mode 100644 index 0000000..5a3c7ef --- /dev/null +++ b/sound/soc/codecs/ha-dsp.c @@ -0,0 +1,333 @@ +/* + * ha-dsp.c -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Authors: + * Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Stefan Roese <sr(a)denx.de> + * Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> + * + * 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. + * + * Maintainer: Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "ha-dsp.h" + +/* Reset default register values for soc-cache */ +static const struct reg_default ha_dsp_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x55 }, + { 0x02, 0x55 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x02 }, + { 0x09, 0x02 }, + { 0x0a, 0x02 }, + { 0x0b, 0x02 }, + { 0x0c, 0x02 }, + { 0x0d, 0x02 }, + { 0x0e, 0x02 }, + { 0x0f, 0x02 }, +}; + +/* DSP mode selection */ +static const char *const ha_dsp_mode_texts[] = {"Mode 1", "Mode 2"}; +static SOC_ENUM_SINGLE_DECL(ha_dsp_mode_enum, HA_DSP_CTRL, 0, + ha_dsp_mode_texts); + +/* Monitor output mux selection */ +static const char *const ha_dsp_monitor_texts[] = {"Off", "ADC", "DAC"}; +static SOC_ENUM_SINGLE_DECL(ha_dsp_monitor_enum, HA_DSP_CTRL, 1, + ha_dsp_monitor_texts); + +static const struct snd_kcontrol_new ha_dsp_monitor_control = + SOC_DAPM_ENUM("Route", ha_dsp_monitor_enum); + +/* Output mixers */ +static const struct snd_kcontrol_new ha_dsp_out1_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT1_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out2_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT2_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT2_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out3_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT3_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT3_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out4_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT4_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT4_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out5_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT5_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT5_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out6_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT6_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT6_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out7_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT7_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out8_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT8_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT8_CTRL, 2, 1, 0), +}; + +static const struct snd_kcontrol_new ha_dsp_snd_controls[] = { + SOC_SINGLE("ADC Capture Volume", + HA_DSP_ADC_VOL, 0, 0x7f, 0), + SOC_SINGLE("ADC Capture Switch", + HA_DSP_ADC_VOL, 7, 0x01, 1), + + SOC_SINGLE("PCM Playback Volume", + HA_DSP_DAC_VOL, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", + HA_DSP_DAC_VOL, 7, 0x01, 1), + + SOC_ENUM("DSP Mode", ha_dsp_mode_enum), +}; + +static const struct snd_soc_dapm_widget ha_dsp_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("OUT1 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out1_mixer_controls), + SOC_MIXER_ARRAY("OUT2 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out2_mixer_controls), + SOC_MIXER_ARRAY("OUT3 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out3_mixer_controls), + SOC_MIXER_ARRAY("OUT4 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out4_mixer_controls), + SOC_MIXER_ARRAY("OUT5 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out5_mixer_controls), + SOC_MIXER_ARRAY("OUT6 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out6_mixer_controls), + SOC_MIXER_ARRAY("OUT7 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out7_mixer_controls), + SOC_MIXER_ARRAY("OUT8 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out8_mixer_controls), + + SND_SOC_DAPM_PGA("OUT1 PGA", HA_DSP_OUT1_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT2 PGA", HA_DSP_OUT2_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT3 PGA", HA_DSP_OUT3_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT4 PGA", HA_DSP_OUT4_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT5 PGA", HA_DSP_OUT5_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT6 PGA", HA_DSP_OUT6_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT7 PGA", HA_DSP_OUT7_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT8 PGA", HA_DSP_OUT8_CTRL, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Monitor Out Mux", SND_SOC_NOPM, 0, 0, + &ha_dsp_monitor_control), + + /* Input pins */ + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN7"), + SND_SOC_DAPM_INPUT("IN8"), + + /* Output pins */ + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("OUT4"), + SND_SOC_DAPM_OUTPUT("OUT5"), + SND_SOC_DAPM_OUTPUT("OUT6"), + SND_SOC_DAPM_OUTPUT("OUT7"), + SND_SOC_DAPM_OUTPUT("OUT8"), + SND_SOC_DAPM_OUTPUT("MONITOR"), +}; + +static const struct snd_soc_dapm_route ha_dsp_routes[] = { + /* Inputs to ADC */ + {"ADC", NULL, "IN1"}, + {"ADC", NULL, "IN2"}, + {"ADC", NULL, "IN3"}, + {"ADC", NULL, "IN4"}, + {"ADC", NULL, "IN5"}, + {"ADC", NULL, "IN6"}, + {"ADC", NULL, "IN7"}, + {"ADC", NULL, "IN8"}, + + /* DAC and input bypass paths to outputs */ + {"OUT1 Mixer", "DAC Switch", "DAC"}, + {"OUT1 Mixer", "IN Bypass Switch", "IN1"}, + {"OUT1 PGA", NULL, "OUT1 Mixer"}, + {"OUT1", NULL, "OUT1 PGA"}, + + {"OUT2 Mixer", "DAC Switch", "DAC"}, + {"OUT2 Mixer", "IN Bypass Switch", "IN2"}, + {"OUT2 PGA", NULL, "OUT2 Mixer"}, + {"OUT2", NULL, "OUT2 PGA"}, + + {"OUT3 Mixer", "DAC Switch", "DAC"}, + {"OUT3 Mixer", "IN Bypass Switch", "IN3"}, + {"OUT3 PGA", NULL, "OUT3 Mixer"}, + {"OUT3", NULL, "OUT3 PGA"}, + + {"OUT4 Mixer", "DAC Switch", "DAC"}, + {"OUT4 Mixer", "IN Bypass Switch", "IN4"}, + {"OUT4 PGA", NULL, "OUT4 Mixer"}, + {"OUT4", NULL, "OUT4 PGA"}, + + {"OUT5 Mixer", "DAC Switch", "DAC"}, + {"OUT5 Mixer", "IN Bypass Switch", "IN5"}, + {"OUT5 PGA", NULL, "OUT5 Mixer"}, + {"OUT5", NULL, "OUT5 PGA"}, + + {"OUT6 Mixer", "DAC Switch", "DAC"}, + {"OUT6 Mixer", "IN Bypass Switch", "IN6"}, + {"OUT6 PGA", NULL, "OUT6 Mixer"}, + {"OUT6", NULL, "OUT6 PGA"}, + + {"OUT7 Mixer", "DAC Switch", "DAC"}, + {"OUT7 Mixer", "IN Bypass Switch", "IN7"}, + {"OUT7 PGA", NULL, "OUT7 Mixer"}, + {"OUT7", NULL, "OUT7 PGA"}, + + {"OUT8 Mixer", "DAC Switch", "DAC"}, + {"OUT8 Mixer", "IN Bypass Switch", "IN8"}, + {"OUT8 PGA", NULL, "OUT8 Mixer"}, + {"OUT8", NULL, "OUT8 PGA"}, + + /* Monitor output */ + {"Monitor Out Mux", "ADC", "ADC"}, + {"Monitor Out Mux", "DAC", "DAC"}, + {"MONITOR", NULL, "Monitor Out Mux"}, +}; + +static struct snd_soc_dai_driver ha_dsp_dai = { + .name = "ha-dsp-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_96000, + /* We use only 32 Bits for Audio */ + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_96000, + /* We use only 32 Bits for Audio */ + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, +}; + +static int ha_dsp_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->control_data = dev_get_regmap(codec->dev, NULL); + ret = snd_soc_codec_set_cache_io(codec, codec->control_data); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET); + + return 0; +} + +static int ha_dsp_remove(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ha_dsp = { + .probe = ha_dsp_probe, + .remove = ha_dsp_remove, + + .controls = ha_dsp_snd_controls, + .num_controls = ARRAY_SIZE(ha_dsp_snd_controls), + .dapm_widgets = ha_dsp_widgets, + .num_dapm_widgets = ARRAY_SIZE(ha_dsp_widgets), + .dapm_routes = ha_dsp_routes, + .num_dapm_routes = ARRAY_SIZE(ha_dsp_routes), +}; + +static const struct regmap_config ha_dsp_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = 0x0f, + .reg_defaults = ha_dsp_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ha_dsp_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int ha_dsp_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(client, &ha_dsp_regmap); + if (IS_ERR(regmap)) { + dev_err(&client->dev, "Failed to create regmap: %ld\n", + PTR_ERR(regmap)); + return PTR_ERR(regmap); + } + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ha_dsp, + &ha_dsp_dai, 1); +} + +static int ha_dsp_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +/* + * This name/ID is neded to match the DT node for the codec + */ +static const struct i2c_device_id ha_dsp_i2c_id[] = { + { "ha-dsp-audio", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ha_dsp_i2c_id); + +static struct i2c_driver ha_dsp_i2c_driver = { + .driver = { + .name = "ha-dsp-codec", + .owner = THIS_MODULE, + }, + .probe = ha_dsp_i2c_probe, + .remove = ha_dsp_i2c_remove, + .id_table = ha_dsp_i2c_id, +}; + +module_i2c_driver(ha_dsp_i2c_driver); + +MODULE_DESCRIPTION("ASoC HA DSP driver"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ha-dsp.h b/sound/soc/codecs/ha-dsp.h new file mode 100644 index 0000000..6622f8a --- /dev/null +++ b/sound/soc/codecs/ha-dsp.h @@ -0,0 +1,50 @@ +/* + * ha-dsp.h -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Author: Jarkko Nikula <jhnikula(a)gmail.com> + * + * 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 __HA_DSP_H__ +#define __HA_DSP_H__ + +/* Registers */ + +/* + * Bit 2-1: Monitor output selection: Off, ADC, DAC + * Bit 0: DSP Mode + */ +#define HA_DSP_CTRL 0x00 + +/* + * Bit 7: Mute + * Bit 6-0: Volume + */ +#define HA_DSP_DAC_VOL 0x01 +#define HA_DSP_ADC_VOL 0x02 + +/* + * Bit 2: INx Bypass to OUTx Switch + * Bit 1: DAC to OUTx switch + * Bit 0: Output power + */ +#define HA_DSP_OUT1_CTRL 0x08 +#define HA_DSP_OUT2_CTRL 0x09 +#define HA_DSP_OUT3_CTRL 0x0a +#define HA_DSP_OUT4_CTRL 0x0b +#define HA_DSP_OUT5_CTRL 0x0c +#define HA_DSP_OUT6_CTRL 0x0d +#define HA_DSP_OUT7_CTRL 0x0e +#define HA_DSP_OUT8_CTRL 0x0f + +/* Register bits and values */ + +/* HA_DSP_CTRL */ +#define HA_DSP_SW_RESET 0xff + +#endif -- 1.9.2

2 3

[alsa-devel] [PATCH v3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 16 Aug '14

16 Aug '14

This patch adds a CODEC function to the NXP TDA998x HDMI transmitter. The CODEC handles both I2S and S/PDIF input and does dynamic input switch in the TDA998x I2C driver on start/stop audio streaming. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) --- .../devicetree/bindings/drm/i2c/tda998x.txt | 13 ++ drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/tda998x_codec.c | 247 +++++++++++++++++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 74 ++++-- drivers/gpu/drm/i2c/tda998x_drv.h | 32 +++ include/drm/i2c/tda998x.h | 1 + 6 files changed, 348 insertions(+), 21 deletions(-) create mode 100644 drivers/gpu/drm/i2c/tda998x_codec.c create mode 100644 drivers/gpu/drm/i2c/tda998x_drv.h diff --git a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt index d7df01c..7ce4eac 100644 --- a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt +++ b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt @@ -15,6 +15,15 @@ Optional properties: - video-ports: 24 bits value which defines how the video controller output is wired to the TDA998x input - default: <0x230145> + - audio-ports: one or two values corresponding to entries in + the audio-port-names property. + + - audio-port-names: must contain "i2s", "spdif" entries + matching entries in the audio-ports property. + + - #sound-dai-cells: must be set to <1> for use with the simple-card. + The DAI 0 is the I2S input and the DAI 1 is the S/PDIF input. + Example: tda998x: hdmi-encoder { @@ -24,4 +33,8 @@ Example: interrupts = <27 2>; /* falling edge */ pinctrl-0 = <&pmx_camera>; pinctrl-names = "default"; + + audio-ports = <0x03>, <0x04>; + audio-port-names = "i2s", "spdif"; + #sound-dai-cells = <1>; }; diff --git a/drivers/gpu/drm/i2c/Makefile b/drivers/gpu/drm/i2c/Makefile index 43aa33b..f2d625c 100644 --- a/drivers/gpu/drm/i2c/Makefile +++ b/drivers/gpu/drm/i2c/Makefile @@ -6,5 +6,5 @@ obj-$(CONFIG_DRM_I2C_CH7006) += ch7006.o sil164-y := sil164_drv.o obj-$(CONFIG_DRM_I2C_SIL164) += sil164.o -tda998x-y := tda998x_drv.o +tda998x-y := tda998x_drv.o tda998x_codec.o obj-$(CONFIG_DRM_I2C_NXP_TDA998X) += tda998x.o diff --git a/drivers/gpu/drm/i2c/tda998x_codec.c b/drivers/gpu/drm/i2c/tda998x_codec.c new file mode 100644 index 0000000..e22f974 --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_codec.c @@ -0,0 +1,247 @@ +/* + * ALSA SoC TDA998X CODEC + * + * Copyright (C) 2014 Jean-Francois Moine + * + * 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 <linux/module.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <linux/of.h> +#include <linux/i2c.h> +#include <drm/drm_encoder_slave.h> +#include <drm/i2c/tda998x.h> + +#include "tda998x_drv.h" + +#define TDA998X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int tda_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + u8 *eld = priv->eld; + struct snd_pcm_runtime *runtime = substream->runtime; + u8 *sad; + int sad_count; + unsigned eld_ver, mnl, rate_mask; + unsigned max_channels, fmt; + u64 formats; + struct snd_pcm_hw_constraint_list *rate_constraints = + &priv->rate_constraints; + static const u32 hdmi_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000 + }; + + /* check if streaming is already active */ + if (priv->dai_id != AFMT_NO_AUDIO) + return -EBUSY; + priv->dai_id = dai->id; + + if (!eld) + return 0; + + /* adjust the hw params from the ELD (EDID) */ + eld_ver = eld[0] >> 3; + if (eld_ver != 2 && eld_ver != 31) + return 0; + + mnl = eld[4] & 0x1f; + if (mnl > 16) + return 0; + + sad_count = eld[5] >> 4; + sad = eld + 20 + mnl; + + /* Start from the basic audio settings */ + max_channels = 2; + rate_mask = 0; + fmt = 0; + while (sad_count--) { + switch (sad[0] & 0x78) { + case 0x08: /* PCM */ + max_channels = max(max_channels, (sad[0] & 7) + 1u); + rate_mask |= sad[1]; + fmt |= sad[2] & 0x07; + break; + } + sad += 3; + } + + /* set the constraints */ + rate_constraints->list = hdmi_rates; + rate_constraints->count = ARRAY_SIZE(hdmi_rates); + rate_constraints->mask = rate_mask; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + rate_constraints); + + formats = 0; + if (fmt & 1) + formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (fmt & 2) + formats |= SNDRV_PCM_FMTBIT_S20_3LE; + if (fmt & 4) + formats |= SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_mask64(runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + formats); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, max_channels); + return 0; +} + +static int tda_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + /* Requires an attached display */ + if (!priv->encoder->crtc) + return -ENODEV; + + /* if same input and same parameters, do not do a full switch */ + if (dai->id == priv->params.audio_format && + params_format(params) == priv->audio_sample_format) { + tda998x_audio_start(priv, 0); + return 0; + } + priv->params.audio_format = dai->id; + priv->audio_sample_format = params_format(params); + priv->params.audio_cfg = + priv->audio_ports[dai->id == AFMT_I2S ? 0 : 1]; + tda998x_audio_start(priv, 1); + return 0; +} + +static void tda_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + tda998x_audio_stop(priv); + priv->dai_id = AFMT_NO_AUDIO; +} + +static const struct snd_soc_dai_ops tda_ops = { + .startup = tda_startup, + .hw_params = tda_hw_params, + .shutdown = tda_shutdown, +}; + +static struct snd_soc_dai_driver tda998x_dai[] = { + { + .name = "i2s-hifi", + .id = AFMT_I2S, + .playback = { + .stream_name = "HDMI I2S Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, + { + .name = "spdif-hifi", + .id = AFMT_SPDIF, + .playback = { + .stream_name = "HDMI SPDIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 22050, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, +}; + +static const struct snd_soc_dapm_widget tda_widgets[] = { + SND_SOC_DAPM_OUTPUT("hdmi-out"), +}; +static const struct snd_soc_dapm_route tda_routes[] = { + { "hdmi-out", NULL, "HDMI I2S Playback" }, + { "hdmi-out", NULL, "HDMI SPDIF Playback" }, +}; + +static int tda_probe(struct snd_soc_codec *codec) +{ + struct i2c_client *i2c_client = to_i2c_client(codec->dev); + struct tda998x_priv *priv = i2c_get_clientdata(i2c_client); + struct device_node *np = codec->dev->of_node; + int i, j, ret; + const char *p; + + if (!priv) + return -ENODEV; + snd_soc_codec_set_drvdata(codec, priv); + + if (!np) + return 0; + + /* get the audio input ports*/ + for (i = 0; i < 2; i++) { + u32 port; + + ret = of_property_read_u32_index(np, "audio-ports", i, &port); + if (ret) { + if (i == 0) + dev_err(codec->dev, + "bad or missing audio-ports\n"); + break; + } + ret = of_property_read_string_index(np, "audio-port-names", + i, &p); + if (ret) { + dev_err(codec->dev, + "missing audio-port-names[%d]\n", i); + break; + } + if (strcmp(p, "i2s") == 0) { + j = 0; + } else if (strcmp(p, "spdif") == 0) { + j = 1; + } else { + dev_err(codec->dev, + "bad audio-port-names '%s'\n", p); + break; + } + priv->audio_ports[j] = port; + } + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_tda998x = { + .probe = tda_probe, + .dapm_widgets = tda_widgets, + .num_dapm_widgets = ARRAY_SIZE(tda_widgets), + .dapm_routes = tda_routes, + .num_dapm_routes = ARRAY_SIZE(tda_routes), +}; + +int tda998x_codec_register(struct device *dev) +{ + return snd_soc_register_codec(dev, + &soc_codec_tda998x, + tda998x_dai, ARRAY_SIZE(tda998x_dai)); +} + +void tda998x_codec_unregister(struct device *dev) +{ + snd_soc_unregister_codec(dev); +} diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index 240c331..7e9f9dc 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/irq.h> #include <sound/asoundef.h> +#include <sound/pcm_params.h> #include <drm/drmP.h> #include <drm/drm_crtc_helper.h> @@ -28,24 +29,9 @@ #include <drm/drm_edid.h> #include <drm/i2c/tda998x.h> -#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) +#include "tda998x_drv.h" -struct tda998x_priv { - struct i2c_client *cec; - struct i2c_client *hdmi; - uint16_t rev; - uint8_t current_page; - int dpms; - bool is_hdmi_sink; - u8 vip_cntrl_0; - u8 vip_cntrl_1; - u8 vip_cntrl_2; - struct tda998x_encoder_params params; - - wait_queue_head_t wq_edid; - volatile int wq_edid_wait; - struct drm_encoder *encoder; -}; +#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) #define to_tda998x_priv(x) ((struct tda998x_priv *)to_encoder_slave(x)->slave_priv) @@ -640,12 +626,11 @@ static void tda998x_configure_audio(struct tda998x_priv *priv, struct drm_display_mode *mode, struct tda998x_encoder_params *p) { - uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv; + uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv, aclk; uint32_t n; /* Enable audio ports */ reg_write(priv, REG_ENA_AP, p->audio_cfg); - reg_write(priv, REG_ENA_ACLK, p->audio_clk_cfg); /* Set audio input source */ switch (p->audio_format) { @@ -654,13 +639,28 @@ tda998x_configure_audio(struct tda998x_priv *priv, clksel_aip = AIP_CLKSEL_AIP_SPDIF; clksel_fs = AIP_CLKSEL_FS_FS64SPDIF; cts_n = CTS_N_M(3) | CTS_N_K(3); + aclk = 0; /* no clock */ break; case AFMT_I2S: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_I2S); clksel_aip = AIP_CLKSEL_AIP_I2S; clksel_fs = AIP_CLKSEL_FS_ACLK; - cts_n = CTS_N_M(3) | CTS_N_K(3); + /* with I2S input, the CTS_N predivider depends on + * the sample width */ + switch (priv->audio_sample_format) { + case SNDRV_PCM_FORMAT_S16_LE: + cts_n = CTS_N_M(3) | CTS_N_K(1); + break; + case SNDRV_PCM_FORMAT_S24_LE: + cts_n = CTS_N_M(3) | CTS_N_K(2); + break; + default: + case SNDRV_PCM_FORMAT_S32_LE: + cts_n = CTS_N_M(3) | CTS_N_K(3); + break; + } + aclk = 1; /* clock enable */ break; default: @@ -672,6 +672,7 @@ tda998x_configure_audio(struct tda998x_priv *priv, reg_clear(priv, REG_AIP_CNTRL_0, AIP_CNTRL_0_LAYOUT | AIP_CNTRL_0_ACR_MAN); /* auto CTS */ reg_write(priv, REG_CTS_N, cts_n); + reg_write(priv, REG_ENA_ACLK, aclk); /* * Audio input somehow depends on HDMI line rate which is @@ -728,6 +729,24 @@ tda998x_configure_audio(struct tda998x_priv *priv, tda998x_write_aif(priv, p); } +/* tda998x codec interface */ +void tda998x_audio_start(struct tda998x_priv *priv, + int full) +{ + struct tda998x_encoder_params *p = &priv->params; + + if (!full) { + reg_write(priv, REG_ENA_AP, p->audio_cfg); + return; + } + tda998x_configure_audio(priv, &priv->encoder->crtc->hwmode, p); +} + +void tda998x_audio_stop(struct tda998x_priv *priv) +{ + reg_write(priv, REG_ENA_AP, 0); +} + /* DRM encoder functions */ static void @@ -1149,6 +1168,11 @@ tda998x_encoder_get_modes(struct drm_encoder *encoder, drm_mode_connector_update_edid_property(connector, edid); n = drm_add_edid_modes(connector, edid); priv->is_hdmi_sink = drm_detect_hdmi_monitor(edid); + + /* keep the EDID as ELD for the audio subsystem */ + drm_edid_to_eld(connector, edid); + priv->eld = connector->eld; + kfree(edid); } @@ -1191,6 +1215,8 @@ tda998x_encoder_destroy(struct drm_encoder *encoder) if (priv->hdmi->irq) free_irq(priv->hdmi->irq, priv); + tda998x_codec_unregister(&priv->hdmi->dev); + if (priv->cec) i2c_unregister_device(priv->cec); kfree(priv); @@ -1239,10 +1265,15 @@ tda998x_encoder_init(struct i2c_client *client, if (!priv) return -ENOMEM; + i2c_set_clientdata(client, priv); + priv->vip_cntrl_0 = VIP_CNTRL_0_SWAP_A(2) | VIP_CNTRL_0_SWAP_B(3); priv->vip_cntrl_1 = VIP_CNTRL_1_SWAP_C(0) | VIP_CNTRL_1_SWAP_D(1); priv->vip_cntrl_2 = VIP_CNTRL_2_SWAP_E(4) | VIP_CNTRL_2_SWAP_F(5); + priv->params.audio_frame[1] = 1; /* channels - 1 */ + priv->params.audio_sample_rate = 48000; /* 48kHz */ + priv->current_page = 0xff; priv->hdmi = client; priv->cec = i2c_new_dummy(client->adapter, 0x34); @@ -1340,6 +1371,9 @@ tda998x_encoder_init(struct i2c_client *client, /* enable EDID read irq: */ reg_set(priv, REG_INT_FLAGS_2, INT_FLAGS_2_EDID_BLK_RD); + /* register the audio CODEC */ + tda998x_codec_register(&client->dev); + if (!np) return 0; /* non-DT */ diff --git a/drivers/gpu/drm/i2c/tda998x_drv.h b/drivers/gpu/drm/i2c/tda998x_drv.h new file mode 100644 index 0000000..e6c8dd5 --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_drv.h @@ -0,0 +1,32 @@ +/* tda998x private data */ + +struct tda998x_priv { + struct i2c_client *cec; + struct i2c_client *hdmi; + uint16_t rev; + uint8_t current_page; + int dpms; + bool is_hdmi_sink; + u8 vip_cntrl_0; + u8 vip_cntrl_1; + u8 vip_cntrl_2; + struct tda998x_encoder_params params; + + wait_queue_head_t wq_edid; + volatile int wq_edid_wait; + struct drm_encoder *encoder; + + u8 audio_ports[2]; + int audio_sample_format; + int dai_id; /* DAI ID when streaming active */ + + u8 *eld; + + struct snd_pcm_hw_constraint_list rate_constraints; +}; + +int tda998x_codec_register(struct device *dev); +void tda998x_codec_unregister(struct device *dev); + +void tda998x_audio_start(struct tda998x_priv *priv, int full); +void tda998x_audio_stop(struct tda998x_priv *priv); diff --git a/include/drm/i2c/tda998x.h b/include/drm/i2c/tda998x.h index 3e419d9..31757df 100644 --- a/include/drm/i2c/tda998x.h +++ b/include/drm/i2c/tda998x.h @@ -20,6 +20,7 @@ struct tda998x_encoder_params { u8 audio_frame[6]; enum { + AFMT_NO_AUDIO = 0, AFMT_SPDIF, AFMT_I2S } audio_format; -- 2.0.1

3 2

[alsa-devel] [PATCH RFC] ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support
by Nicolin Chen 16 Aug '14

16 Aug '14

The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that can be used, ideally, for all Freescale CPU DAI drivers and external CODECs. The idea of this generic sound card is a bit like ASoC Simple Card. However, for Freescale SoCs (especially those released in recent years), most of them have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And this is a specific feature that might be painstakingly controlled and merged into the Simple Card driver. So having this driver will allow all Freescale SoC users to benefit from the simplification to support a new card and the capability of wide sample rates support through ASRC. The driver is initially designed for sound card using I2S or PCM DAI formats. However, it's also possible to merge those non-I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, into this card as long as the merge will not break the original function and as long as there is something redundant that can be abstracted along with I2S type sound cards. As an initial version, it only supports three cards that I can test: imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver imx-audio-wm8962, just like the old imx-wm8962.c driver The driver is also compatible with the old Device Tree bindings of WM8962 and SGTL5000. So we may consider to remove those two drivers after this driver is totally enabled. (It needs to be added into defconfig) Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 82 +++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 573 +++++++++++++++++++++ 4 files changed, 673 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.txt create mode 100644 sound/soc/fsl/fsl-asoc-card.c diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt new file mode 100644 index 0000000..a96774c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -0,0 +1,82 @@ +Freescale Generic ASoC Sound Card with ASRC support + +The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale +SoCs connecting with external CODECs. + +The idea of this generic sound card is a bit like ASoC Simple Card. However, +for Freescale SoCs (especially those released in recent years), most of them +have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And +this is a specific feature that might be painstakingly controlled and merged +into the Simple Card. + +So having this generic sound card allows all Freescale SoC users to benefit +from the simplification of a new card support and the capability of the wide +sample rates support through ASRC. + +Note: The card is initially designed for those sound cards who use I2S and + PCM DAI formats. However, it'll be also possible to support those non + I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long + as the driver has been properly upgraded. + + +The compatible list for this generic sound card currently: + "fsl,imx-audio-cs42888" + + "fsl,imx-audio-wm8962" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) + + "fsl,imx-audio-sgtl5000" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) + +Required properties: + + - compatible : Contains one of entries in the compatible list. + + - model : The user-visible name of this sound complex + + - audio-cpu : The phandle of an CPU DAI controller + + - audio-codec : The phandle of an audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. There're a few pre-designed board connectors: + * Line Out Jack + * Line In Jack + * Headphone Jack + * Mic Jack + * Ext Spk + * AMIC (stands for Analog Microphone Jack) + * DMIC (stands for Digital Microphone Jack) + + Note: The "Mic Jack" and "AMIC" are redundant while + coexsiting in order to support the old bindings + of wm8962 and sgtl5000. + +Optional properties: + + - audio-asrc : The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + +Example: +sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 2fb8a43..b55bf36 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -58,6 +58,22 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + select SND_SOC_CS42XX8_I2C + select SND_SOC_SGTL5000 + select SND_SOC_WM8962 + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9ff5926..8f6d84e 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o @@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 0000000..cf3f1f4 --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,573 @@ +/* + * Freescale Generic ASoC Sound Card driver with ASRC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <nicoleotsuka(a)gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "fsl_esai.h" +#include "fsl_sai.h" +#include "imx-audmux.h" + +#include "../codecs/sgtl5000.h" +#include "../codecs/wm8962.h" + +#define RX 0 +#define TX 1 + +/* Default DAI format without Master and Slave flag */ +#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) + +/** + * CODEC private data + * + * @mclk_freq: Clock rate of MCLK + * @mclk_id: MCLK (or main clock) id for set_sysclk() + * @fll_id: FLL (or secordary clock) id for set_sysclk() + * @pll_id: PLL id for set_pll() + */ +struct codec_priv { + unsigned long mclk_freq; + u32 mclk_id; + u32 fll_id; + u32 pll_id; +}; + +/** + * CPU private data + * + * @sysclk_freq[2]: SYSCLK rates for set_sysclk() + * @sysclk_dir[2]: SYSCLK directions for set_sysclk() + * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * + * Note: [1] for tx and [0] for rx + */ +struct cpu_priv { + unsigned long sysclk_freq[2]; + u32 sysclk_dir[2]; + u32 sysclk_id[2]; +}; + +/** + * Freescale Generic ASOC card private data + * + * @dai_link[3]: DAI link structure including normal one and DPCM link + * @pdev: platform device pointer + * @codec_priv: CODEC private data + * @cpu_priv: CPU private data + * @card: ASoC card structure + * @sample_rate: Current sample rate + * @sample_format: Current sample format + * @asrc_rate: ASRC sample rate used by Back-Ends + * @asrc_format: ASRC sample format used by Back-Ends + * @dai_fmt: DAI format between CPU and CODEC + * @name: Card name + */ + +struct fsl_asoc_card_priv { + struct snd_soc_dai_link dai_link[3]; + struct platform_device *pdev; + struct codec_priv codec_priv; + struct cpu_priv cpu_priv; + struct snd_soc_card card; + u32 sample_rate; + u32 sample_format; + u32 asrc_rate; + u32 asrc_format; + u32 dai_fmt; + char name[32]; +}; + +/** + * This dapm route map exsits for DPCM link only. + * The other routes shall go through Device Tree. + */ +static const struct snd_soc_dapm_route audio_map[] = { + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "Capture"}, +}; + +/* Add all possible widgets into here without being redundant */ +static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct cpu_priv *cpu_priv = &priv->cpu_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); + + if (priv->card.set_bias_level) + return 0; + + /* Specific configurations of DAIs starts from here */ + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + cpu_priv->sysclk_freq[tx], + cpu_priv->sysclk_dir[tx]); + if (ret) { + dev_err(dev, "failed to set sysclk for cpu dai\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops fsl_asoc_card_ops = { + .hw_params = fsl_asoc_card_hw_params, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_interval *rate; + struct snd_mask *mask; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = rate->min = priv->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, priv->asrc_format); + + return 0; +} + +static struct snd_soc_dai_link fsl_asoc_card_dai[] = { + /* Default ASoC DAI Link*/ + { + .name = "HiFi", + .stream_name = "HiFi", + .ops = &fsl_asoc_card_ops, + }, + /* DPCM Link between Front-End and Back-End (Optional) */ + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .dpcm_capture = 1, + .dynamic = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .platform_name = "snd-soc-dummy", + .be_hw_params_fixup = be_hw_params_fixup, + .ops = &fsl_asoc_card_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level != SND_SOC_BIAS_STANDBY) + break; + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_PREPARE) + break; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); + if (ret) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + return 0; +} + +static int fsl_asoc_card_audmux_init(struct device_node *np, + struct fsl_asoc_card_priv *priv) +{ + struct device *dev = &priv->pdev->dev; + u32 int_ptcr = 0, ext_ptcr = 0; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the AUDMUX API expects it starts at 0. + */ + int_port--; + ext_port--; + + /* + * Use asynchronous mode (6 wires) for all cases. + * If only 4 wires are needed, just set SSI into + * synchronous mode and enable 4 PADs in IOMUX. + */ + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + default: + return -EINVAL; + } + + /* Asynchronous mode can not be set along with RCLKDIR */ + ret = imx_audmux_v2_configure_port(int_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(int_port, int_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_late_probe(struct snd_soc_card *card) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *asrc_pdev = NULL; + struct platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct i2c_client *codec_dev; + struct clk *codec_clk; + u32 width; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + cpu_np = of_parse_phandle(np, "audio-cpu", 0); + /* Give a chance to old DT binding */ + if (!cpu_np) + cpu_np = of_parse_phandle(np, "ssi-controller", 0); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!cpu_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find CPU DAI device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + asrc_np = of_parse_phandle(np, "audio-asrc", 0); + if (asrc_np) + asrc_pdev = of_find_device_by_node(asrc_np); + + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ + codec_clk = clk_get(&codec_dev->dev, NULL); + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } + + /* Default sample rate and format, will be updated in hw_params() */ + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; + + /* Assign a default DAI format, and allow each card to overwrite it */ + priv->dai_fmt = DAI_FMT_BASE; + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + priv->card.set_bias_level = NULL; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + priv->codec_priv.mclk_id = SGTL5000_SYSCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv.pll_id = WM8962_FLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + return -EINVAL; + } + + /* Common settings for corresponding Freescale CPU DAI driver */ + if (strstr(cpu_np->name, "ssi")) { + /* Only SSI needs to configure AUDMUX */ + ret = fsl_asoc_card_audmux_init(np, priv); + if (ret) { + dev_err(&pdev->dev, "failed to init audmux\n"); + goto fail; + } + } else if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; + priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; + } else if (strstr(cpu_np->name, "sai")) { + priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; + priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; + } + + sprintf(priv->name, "%s-audio", codec_dev->name); + + /* Initialize sound card */ + priv->pdev = pdev; + priv->card.dev = &pdev->dev; + priv->card.name = priv->name; + priv->card.dai_link = priv->dai_link; + priv->card.dapm_routes = audio_map; + priv->card.late_probe = fsl_asoc_card_late_probe; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); + priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + + memcpy(priv->dai_link, fsl_asoc_card_dai, + sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + + /* Normal DAI Link */ + priv->dai_link[0].cpu_of_node = cpu_np; + priv->dai_link[0].codec_of_node = codec_np; + priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].platform_of_node = cpu_np; + priv->dai_link[0].dai_fmt = priv->dai_fmt; + priv->card.num_links = 1; + + if (asrc_pdev) { + /* DPCM DAI Links only if ASRC exsits */ + priv->dai_link[1].cpu_of_node = asrc_np; + priv->dai_link[1].platform_of_node = asrc_np; + priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].cpu_of_node = cpu_np; + priv->dai_link[2].dai_fmt = priv->dai_fmt; + priv->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + if (width == 24) + priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + + /* Finish card registering */ + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + +fail: + of_node_put(codec_np); + of_node_put(asrc_np); + of_node_put(cpu_np); + + return ret; +} + +static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8962", }, + {} +}; + +static struct platform_driver fsl_asoc_card_driver = { + .probe = fsl_asoc_card_probe, + .driver = { + .name = "fsl-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = fsl_asoc_card_dt_ids, + }, +}; +module_platform_driver(fsl_asoc_card_driver); + +MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); +MODULE_AUTHOR("Nicolin Chen <nicoleotsuka(a)gmail.com>"); +MODULE_ALIAS("platform:fsl-asoc-card"); +MODULE_LICENSE("GPL"); -- 1.8.4

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[alsa-devel] [PATCH] ASoC: rt5645: add clock source selection controls
by bardliao＠realtek.com 16 Aug '14

16 Aug '14

From: Bard Liao <bardliao(a)realtek.com> We can select the clock source of some digital widgets in rt5645. This patch adds the controls of those selections. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 97 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 97 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7762d0..e8e30b7 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -440,6 +440,89 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static const char * const rt5645_asrc_clk_source[] = { + "clk_sysy_div_out", "clk_i2s1_track", "clk_i2s2_track", + "clk_i2s3_track", "clk_i2s4_track", "clk_sys2", "clk_sys3", + "clk_sys4", "clk_sys5" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5645_da_sto_asrc_enum, RT5645_ASRC_2, + 12, rt5645_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5645_da_monol_asrc_enum, RT5645_ASRC_2, + 8, rt5645_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5645_da_monor_asrc_enum, RT5645_ASRC_2, + 4, rt5645_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5645_ad_sto1_asrc_enum, RT5645_ASRC_2, + 0, rt5645_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5645_ad_monol_asrc_enum, RT5645_ASRC_3, + 4, rt5645_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5645_ad_monor_asrc_enum, RT5645_ASRC_3, + 0, rt5645_asrc_clk_source); + +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -511,6 +594,20 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { SOC_SINGLE("TDM IF1_DAC1_R Sel", RT5645_TDM_CTRL_3, 8, 7, 0), SOC_SINGLE("TDM IF1_DAC2_L Sel", RT5645_TDM_CTRL_3, 4, 7, 0), SOC_SINGLE("TDM IF1_DAC2_R Sel", RT5645_TDM_CTRL_3, 0, 7, 0), + + /* Clock Source Selection*/ + SOC_ENUM_EXT("DA STO Clk Sel", rt5645_da_sto_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), + SOC_ENUM_EXT("DA MONOL Clk Sel", rt5645_da_monol_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), + SOC_ENUM_EXT("DA MONOR Clk Sel", rt5645_da_monor_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), + SOC_ENUM_EXT("AD STO1 Clk Sel", rt5645_ad_sto1_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), + SOC_ENUM_EXT("AD MONOL Clk Sel", rt5645_ad_monol_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), + SOC_ENUM_EXT("AD MONOR Clk Sel", rt5645_ad_monor_asrc_enum, + snd_soc_get_enum_double, rt5645_clk_sel_put), }; /** -- 1.8.1.1.439.g50a6b54

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[alsa-devel] A minimal playback program does't work on my system
by Carpon frédéric 16 Aug '14

16 Aug '14

Hello, Yesterday i have written. I answer to everybody. I am new and I don't know if there is special rule. Note : It would be good to have a forum to see the messages and answers in a pretty form. the programme in c comes from http://equalarea.com/paul/alsa-audio.html you can download it at https://mon-partage.fr/f/uvyw39wo/ To answer to Michael i have sound in my HDMI with that : >speaker-test -c2 -D hw:0,0 so i did that >gcc -o minimal_playback minimal_playback.c -lasound (warning comes) and >./minimal_playback "plughw:0,0" and i have this result : >Error of segmentation Where i can an exemple with the possibility to do something like that : >./minimal_playback "plughw:0,0" "mywave.wav" Thank you for your help

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[alsa-devel] Alsa on Mac os x
by Carpon frédéric 16 Aug '14

16 Aug '14

Hello , i have looked for an installation of alsa on mac but i don't see. I am afraid if the package Linux would destroy fonctionality with his makefile. Do you know a tutorial to install it, configure for mac ? Thank You for your help

2 1