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

[alsa-devel] [PATCH - pcm_jack plugin 1/1] Restore PID to jack client name when not specified explicitly
by branan＠gmail.com 12 Aug '14

12 Aug '14

From: Branan Purvine-Riley <branan(a)gmail.com> Commit 14190cd8 added a facility to explicitly name Jack clients with a 'name' option, but did so in a way that was not fully backwards-compatible with older versions of the alsa-plugins. There were two issues: 1. Re-using the magical "name" variable, which comes preloaded with the name of the PCM 2. The code assumed snd_config_get_string sets a variable to NULL if the config entry does not exist. Instead, this function leaves the variable unchanged. The result of these two errors meant that the PCM name would be treated as if it were a user-defined name, and the old code which embedded the PID would never be run This code uses a completely new variable to store the explicitly-set client name, so that the two code paths no longer collide. Signed-off-by: Branan Purvine-Riley <branan(a)gmail.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index f5b7129..33f87b1 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -372,6 +372,7 @@ static int make_nonblock(int fd) } static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, + const char* client_name, snd_config_t *playback_conf, snd_config_t *capture_conf, snd_pcm_stream_t stream, int mode) @@ -405,14 +406,14 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -EINVAL; } - if (name == NULL) + if (client_name == NULL) err = snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++); else err = snprintf(jack_client_name, sizeof(jack_client_name), - "%s", name); + "%s", client_name); if (err >= (int)sizeof(jack_client_name)) { fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", @@ -470,6 +471,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_iterator_t i, next; snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; + const char* client_name = NULL; int err; snd_config_for_each(i, next, conf) { @@ -480,7 +482,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; if (strcmp(id, "name") == 0) { - snd_config_get_string(n, &name); + snd_config_get_string(n, &client_name); continue; } if (strcmp(id, "playback_ports") == 0) { @@ -503,7 +505,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, name, client_name, playback_conf, capture_conf, stream, mode); return err; } -- 2.0.4

2 1

[alsa-devel] [PATCH] sound: pci: es1968: use pci_device_id
by Sudip Mukherjee 12 Aug '14

12 Aug '14

Uses the struct pci_device_id instead of the depreciated macro DEFINE_PCI_DEVICE_TABLE. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/es1968.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5bb1cf6..a9956a7 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -570,7 +570,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { +static const struct pci_device_id snd_es1968_ids[] = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ -- 1.8.1.2

2 3

[alsa-devel] [PATCH] ASoC: wm8994: register irq handler for mic1 detect
by Nikesh Oswal 12 Aug '14

12 Aug '14

Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b390..24bcbe3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4099,6 +4099,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) dev_warn(codec->dev, "Failed to request Mic1 detect IRQ: %d\n", ret); + } else { + ret = wm8994_request_irq(wm8994->wm8994, + WM8994_IRQ_MIC1_DET, + wm8994_mic_irq, "Mic 1 detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); } ret = wm8994_request_irq(wm8994->wm8994, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] gpiolib: wm8994: Fix irq offset for gpio_to_irq
by Nikesh Oswal 12 Aug '14

12 Aug '14

Earlier code was using the gpio offset as is to get the irq. For wm8994 driver the gpio related irq's starts from a fixed offset as fixed by this patch. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- drivers/gpio/gpio-wm8994.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/gpio/gpio-wm8994.c b/drivers/gpio/gpio-wm8994.c index d93b6b5..dec47c2 100644 --- a/drivers/gpio/gpio-wm8994.c +++ b/drivers/gpio/gpio-wm8994.c @@ -113,7 +113,8 @@ static int wm8994_gpio_to_irq(struct gpio_chip *chip, unsigned offset) struct wm8994_gpio *wm8994_gpio = to_wm8994_gpio(chip); struct wm8994 *wm8994 = wm8994_gpio->wm8994; - return regmap_irq_get_virq(wm8994->irq_data, offset); + return regmap_irq_get_virq(wm8994->irq_data, + WM8994_IRQ_GPIO(offset + 1)); } -- 1.7.9.5

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[alsa-devel] [PATCHv2] ASOC add wm8973 support to wm8971
by Xavier Hsu 12 Aug '14

12 Aug '14

This patch adds support for the wm8973 codec. After the initial comments, we merged wm8973 driver into wm8971 driver. Any comments about improving the patch are welcome. Thanks. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> Signed-off-by: Andy Green <andy.green(a)linaro.org> --- Documentation/devicetree/bindings/sound/wm8971.txt | 26 + sound/soc/codecs/wm8971.c | 777 +++++++++++++++++--- sound/soc/codecs/wm8971.h | 21 +- 3 files changed, 696 insertions(+), 128 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8971.txt diff --git a/Documentation/devicetree/bindings/sound/wm8971.txt b/Documentation/devicetree/bindings/sound/wm8971.txt new file mode 100644 index 0000000..7313dd1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8971.txt @@ -0,0 +1,26 @@ +WM8971 and WM8973 audio CODEC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8971". + "wlf,wm8973". + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + + - mclk-div : Setting the CLKDIV2 bit for dividing MCLK. + mclk-div = <0> (Default & not divide). + mclk-div = <1> (Divide by 2). + +Example: + +codec: wm8973@1a { + compatible = "wlf,wm8973"; + reg = <0x1a>; + mclk-div = <1>; +}; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 09b7b42..6b1838c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -7,6 +7,10 @@ * * Based on wm8753.c by Liam Girdwood * + * WM8973 support Copyright (C) 2014 Linaro, Ltd + * Author: Xavier Hsu <xavier.hsu(a)linaro.org> + * Andy Green <andy.green(a)linaro.org> + * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your @@ -26,16 +30,31 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "wm8971.h" -#define WM8971_REG_COUNT 43 +enum variant { + VARIANT_WM8971, + VARIANT_WM8973 +}; -static struct workqueue_struct *wm8971_workq = NULL; +#define WM8971_REG_COUNT 43 /* codec private data */ struct wm8971_priv { + struct regmap *regmap; + enum variant variant; unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + int playback_fs; + bool deemph; + int mclk_div; +}; + +static const u16 charge[][2] = { + [VARIANT_WM8971] = { 0xfe3e, 0x01c0 }, + [VARIANT_WM8973] = { 0x003e, 0x01f0 }, }; /* @@ -91,26 +110,53 @@ static const struct reg_default wm8971_reg_defaults[] = { #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) -/* WM8971 Controls */ -static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", - "200Hz @ 48kHz" }; -static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -static const char *wm8971_ng_type[] = { "Constant PGA Gain", - "Mute ADC Output" }; -static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", - "Mono (Right)", "Digital Mono"}; -static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", - "Differential"}; -static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", - "Differential"}; -static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", - "L + R Invert"}; +/* WM8971 / 3 Controls */ +static const char *wm8971_bass[] += { "Linear Control", "Adaptive Boost" }; +static const char *wm8971_bass_filter[] += { "130Hz @ 48kHz", "200Hz @ 48kHz" }; +static const char *wm8971_treble[] += { "8kHz", "4kHz" }; +static const char *wm8971_alc_func[] += { "Off", "Right", "Left", "Stereo" }; +static const char *wm8971_ng_type[] += { "Constant PGA Gain", "Mute ADC Output" }; +static const char *wm8971_deemp[] += { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8971_mono_mux[] += {"Stereo", "Mono (Left)", "Mono (Right)", "Digital Mono"}; + +/* WM8971-only */ +static const char *wm8971_dac_phase[] += { "Non Inverted", "Inverted" }; +static const char *wm8971_lline_mux[] += {"Line", "NC", "NC", "PGA", "Differential"}; +static const char *wm8971_rline_mux[] += {"Line", "Mic", "NC", "PGA", "Differential"}; +static const char *wm8971_lpga_sel[] += {"Line", "NC", "NC", "Differential"}; +static const char *wm8971_rpga_sel[] += {"Line", "Mic", "NC", "Differential"}; +static const char *wm8971_adcpol[] += {"Normal", "L Invert", "R Invert", "L + R Invert"}; + +/* WM8973-only */ +static const char const *wm8973_line_mux[] = {"Line 1", "Line 2", "Line 3", + "PGA", "Differential"}; +static const char const *wm8973_pga_sel[] = {"Line 1", "Line 2", "Line 3", + "Differential"}; +static const char const *wm8973_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut", + "ROUT1"}; +static const char const *wm8973_diff_sel[] = {"Line 1", "Line 2"}; +static const char const *wm8973_adcpol[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static const char const *wm8973_3d_lc[] = { "200Hz", "500Hz" }; +static const char const *wm8973_3d_uc[] = { "2.2kHz", "1.5kHz" }; +static const char const *wm8973_3d_func[] = {"Capture", "Playback"}; + +static const int wm8973_deemph_rates[] = { 0, 32000, 44100, 48000 }; + + static const struct soc_enum wm8971_enum[] = { SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ @@ -136,24 +182,24 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, - WM8971_ROUT1V, 7, 1, 0), + WM8971_ROUT1V, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, - WM8971_ROUT2V, 7, 1, 0), + WM8971_ROUT2V, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, - WM8971_LOUTM2, 4, 7, 1), + WM8971_LOUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, - WM8971_ROUTM2, 4, 7, 1), + WM8971_ROUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, - WM8971_MOUTM2, 4, 7, 1), + WM8971_MOUTM2, 4, 7, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, - WM8971_ROUT1V, 0, 127, 0), + WM8971_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, - WM8971_ROUT2V, 0, 127, 0), + WM8971_ROUT2V, 0, 127, 0), SOC_ENUM("Bass Boost", wm8971_enum[0]), SOC_ENUM("Bass Filter", wm8971_enum[1]), @@ -188,6 +234,224 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), }; +static int wm8973_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + int val = 0, i, best = 0; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8971->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(wm8973_deemph_rates); i++) { + if (abs(wm8973_deemph_rates[i] - wm8971->playback_fs) < + abs(wm8973_deemph_rates[best] - + wm8971->playback_fs)) + best = i; + } + val = best << 1; + } + + dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n", + best, wm8973_deemph_rates[best]); + + return snd_soc_update_bits(codec, WM8971_ADCDAC, 0x6, val); +} + +static int wm8973_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8971->deemph; + + return 0; +} + +static int wm8973_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret = 0; + + if (deemph > 1) + return -EINVAL; + + mutex_lock(&codec->mutex); + if (wm8971->deemph != deemph) { + wm8971->deemph = deemph; + wm8973_set_deemph(codec); + + ret = 1; + } + mutex_unlock(&codec->mutex); + + return ret; +} + +static const DECLARE_TLV_DB_SCALE(in_vol, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(out_vol, -6700, 91, 0); +static const DECLARE_TLV_DB_SCALE(attenuate_6db, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol, -12700, 50, 0); +static const DECLARE_TLV_DB_SCALE(tone_vol, -600, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_tar_vol, -2850, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_max_vol, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol, -9700, 50, 0); +static const DECLARE_TLV_DB_SCALE(bypass_out_vol, -1500, 300, 0); + +static const SOC_ENUM_SINGLE_DECL(bass_boost, WM8971_BASS, 7, wm8971_bass); +static const SOC_ENUM_SINGLE_DECL(bass_filter, WM8971_BASS, + 6, wm8971_bass_filter); +static const SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8971_TREBLE, + 6, wm8971_treble); +static const SOC_ENUM_SINGLE_DECL(lower_cutoff, WM8973_3D, 5, wm8973_3d_lc); +static const SOC_ENUM_SINGLE_DECL(upper_cutoff, WM8973_3D, 6, wm8973_3d_uc); +static const SOC_ENUM_SINGLE_DECL(mode, WM8973_3D, 7, wm8973_3d_func); +static const SOC_ENUM_SINGLE_DECL(alc_capture_func, WM8971_ALC1, + 7, wm8971_alc_func); +static const SOC_ENUM_SINGLE_DECL(alc_capture_ngtype, WM8971_NGATE, + 1, wm8971_ng_type); +static const SOC_ENUM_SINGLE_DECL(left_line, WM8971_LOUTM1, + 0, wm8973_line_mux); +static const SOC_ENUM_SINGLE_DECL(right_line, WM8971_ROUTM1, + 0, wm8973_line_mux); +static const SOC_ENUM_SINGLE_DECL(left_pga, WM8971_LADCIN, 6, wm8973_pga_sel); +static const SOC_ENUM_SINGLE_DECL(right_pga, WM8971_RADCIN, 6, wm8973_pga_sel); +static const SOC_ENUM_SINGLE_DECL(out3, WM8971_ADCTL2, 7, wm8973_out3); +static const SOC_ENUM_SINGLE_DECL(diffmux, WM8971_ADCIN, 8, wm8973_diff_sel); +static const SOC_ENUM_SINGLE_DECL(capture_polarity, WM8971_ADCDAC, + 5, wm8973_adcpol); +static const SOC_ENUM_SINGLE_DECL(monomux, WM8971_ADCIN, 6, wm8971_mono_mux); + + +static const struct snd_kcontrol_new wm8973_snd_controls[] = { + /* Left & Right Input volume */ + SOC_DOUBLE_R_TLV("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, + 0, 63, 0, in_vol), + SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, + 6, 1, 0), + SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), + + /* LOUT1 & ROUT1 volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8971_LOUT1V, + WM8971_ROUT1V, 0, 127, 0, out_vol), + SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, + WM8971_ROUT1V, 7, 1, 0), + + /* ADC & DAC control */ + SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1), + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + wm8973_get_deemph, wm8973_put_deemph), + SOC_ENUM("Capture Polarity", capture_polarity), + SOC_SINGLE_TLV("Playback 6dB Attenuate", WM8971_ADCDAC, + 7, 1, 0, attenuate_6db), + SOC_SINGLE_TLV("Capture 6dB Attenuate", WM8971_ADCDAC, + 8, 1, 0, attenuate_6db), + /* ADCDAC Bit 4 - HPOR */ + + /* Left & Right Channel Digital Volume */ + SOC_DOUBLE_R_TLV("DAC Volume", WM8971_LDAC, WM8971_RDAC, + 0, 255, 0, dac_vol), + + /* Bass Control */ + SOC_SINGLE_TLV("Bass Volume", WM8971_BASS, 0, 15, 1, tone_vol), + SOC_ENUM("Bass Boost", bass_boost), + SOC_ENUM("Bass Filter", bass_filter), + + /* Treble Control */ + SOC_SINGLE_TLV("Treble Volume", WM8971_TREBLE, 0, 15, 0, tone_vol), + SOC_ENUM("Treble Cut-off", treble_cutoff), + + /* 3D Control */ + SOC_SINGLE("3D Switch", WM8973_3D, 0, 1, 0), + SOC_SINGLE("3D Volume", WM8973_3D, 1, 15, 0), + SOC_ENUM("3D Lower Cut-off", lower_cutoff), + SOC_ENUM("3D Upper Cut-off", upper_cutoff), + SOC_ENUM("3D Mode", mode), + + /* ALC1 & ALC2 & ALC3 Control */ + SOC_SINGLE_TLV("ALC Capture Target Volume", WM8971_ALC1, + 0, 15, 0, alc_tar_vol), + SOC_SINGLE_TLV("ALC Capture Max Volume", WM8971_ALC1, + 4, 7, 0, alc_max_vol), + SOC_ENUM("ALC Capture Function", alc_capture_func), + + SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0), + SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0), + + SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0), + + /* Noise Gate Control */ + SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0), + SOC_ENUM("ALC Capture NG Type", alc_capture_ngtype), + SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0), + + /* Left & Right ADC Digital Volume*/ + SOC_DOUBLE_R_TLV("ADC Volume", WM8971_LADC, WM8971_RADC, + 0, 255, 0, adc_vol), + + /* Additional Control 1 */ + SOC_SINGLE("ZC Timeout Switch", WM8971_ADCTL1, 0, 1, 0), + SOC_SINGLE("Playback Invert Switch", WM8971_ADCTL1, 1, 1, 0), + SOC_SINGLE("Analogue Bias", WM8971_ADCTL1, 6, 3, 0), + /* ADCTL1 Bit 2,3 - DATSEL */ + /* ADCTL1 Bit 6,7 - VSEL */ + + /* Additional Control 2 */ + SOC_SINGLE("Right Speaker Playback Invert Switch", WM8971_ADCTL2, + 4, 1, 0), + /* ADCTL2 Bit 2 - LRCM */ + SOC_SINGLE("LRCLK Switch", WM8971_ADCTL2, 2, 1, 0), + /* ADCTL2 Bit 3 - TRI */ + SOC_SINGLE("Headphone Switch POL", WM8971_ADCTL2, 5, 1, 0), + SOC_SINGLE("Headphone Switch EN", WM8971_ADCTL2, 6, 1, 0), + + /* Additional Control 3 */ + /* ADCTL3 Bit 5 - HPFLREN */ + /* ADCTL3 Bit 6 - VROI */ + /* ADCTL3 Bit 7,8 - ADCLRM */ + + /* ADC input Mode */ + /* ADCIN Bit 4 - LDCM */ + /* ADCIN Bit 5 - RDCM */ + /* ADCIN Bit 6,7 - MONOMIX */ + + /* Left & Right ADC Signal Path Control*/ + SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), + + /* Left OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Left Playback Volume", WM8971_LOUTM1, + WM8971_LOUTM2, 4, 7, 1, bypass_out_vol), + + /* Right OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Right Playback Volume", WM8971_ROUTM1, + WM8971_ROUTM2, 4, 7, 1, bypass_out_vol), + + /* Mono OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Mono Playback Volume", WM8971_MOUTM1, + WM8971_MOUTM2, 4, 7, 1, bypass_out_vol), + + /* LOUT2 & ROUT2 volume */ + SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8971_LOUT2V, + WM8971_ROUT2V, 0, 127, 0, out_vol), + SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, + WM8971_ROUT2V, 7, 1, 0), + + /* MONOOUT volume */ + SOC_SINGLE_TLV("Mono Playback Volume", WM8971_MOUTV, + 0, 127, 0, out_vol), + SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), + + SOC_SINGLE("Right Out 2", WM8971_PWR2, 3, 1, 0), + SOC_SINGLE("Left Out 2", WM8971_PWR2, 4, 1, 0), +}; + + /* * DAPM Controls */ @@ -219,18 +483,34 @@ SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0), /* Left Line Mux */ static const struct snd_kcontrol_new wm8971_left_line_controls = SOC_DAPM_ENUM("Route", wm8971_enum[8]); +static const struct snd_kcontrol_new wm8973_left_line_controls = +SOC_DAPM_ENUM("Route", left_line); /* Right Line Mux */ static const struct snd_kcontrol_new wm8971_right_line_controls = SOC_DAPM_ENUM("Route", wm8971_enum[9]); +static const struct snd_kcontrol_new wm8973_right_line_controls = +SOC_DAPM_ENUM("Route", right_line); /* Left PGA Mux */ static const struct snd_kcontrol_new wm8971_left_pga_controls = SOC_DAPM_ENUM("Route", wm8971_enum[10]); +static const struct snd_kcontrol_new wm8973_left_pga_controls = +SOC_DAPM_ENUM("Route", left_pga); /* Right PGA Mux */ static const struct snd_kcontrol_new wm8971_right_pga_controls = SOC_DAPM_ENUM("Route", wm8971_enum[11]); +static const struct snd_kcontrol_new wm8973_right_pga_controls = +SOC_DAPM_ENUM("Route", right_pga); + +/* Out 3 Mux */ +static const struct snd_kcontrol_new wm8973_out3_controls = +SOC_DAPM_ENUM("Route", out3); + +/* Differential Mux */ +static const struct snd_kcontrol_new wm8973_diffmux_controls = +SOC_DAPM_ENUM("Route", diffmux); /* Mono ADC Mux */ static const struct snd_kcontrol_new wm8971_monomux_controls = @@ -238,14 +518,14 @@ SOC_DAPM_ENUM("Route", wm8971_enum[13]); static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_left_mixer_controls[0], - ARRAY_SIZE(wm8971_left_mixer_controls)), + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_right_mixer_controls[0], - ARRAY_SIZE(wm8971_right_mixer_controls)), + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, - &wm8971_mono_mixer_controls[0], - ARRAY_SIZE(wm8971_mono_mixer_controls)), + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), @@ -260,18 +540,18 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, - &wm8971_left_pga_controls), + &wm8971_left_pga_controls), SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, - &wm8971_right_pga_controls), + &wm8971_right_pga_controls), SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_left_line_controls), + &wm8971_left_line_controls), SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_right_line_controls), + &wm8971_right_line_controls), SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("ROUT1"), @@ -284,6 +564,69 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), }; +static const struct snd_soc_dapm_widget wm8973_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), + SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, + &wm8973_right_pga_controls), + SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, + &wm8973_left_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &wm8973_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &wm8973_right_line_controls), + + SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, + &wm8973_out3_controls), + SND_SOC_DAPM_PGA("Out 3", WM8971_PWR2, 1, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &wm8973_diffmux_controls), + + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8971_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8971_monomux_controls), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("MONO1"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("VREF"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("LINPUT3"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT3"), +}; + static const struct snd_soc_dapm_route wm8971_dapm_routes[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, @@ -361,6 +704,85 @@ static const struct snd_soc_dapm_route wm8971_dapm_routes[] = { {"Right ADC", NULL, "Right ADC Mux"}, }; +static const struct snd_soc_dapm_route wm8973_dapm_routes[] = { + {"Left Mixer", "Playback Switch", "Left DAC"}, + {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Left Mixer", "Right Playback Switch", "Right DAC"}, + {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Right Mixer", "Left Playback Switch", "Left DAC"}, + {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Right Mixer", "Playback Switch", "Right DAC"}, + {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Left Out 1", NULL, "Left Mixer"}, + {"LOUT1", NULL, "Left Out 1"}, + + {"Left Out 2", NULL, "Left Mixer"}, + {"LOUT2", NULL, "Left Out 2"}, + + {"Right Out 1", NULL, "Right Mixer"}, + {"ROUT1", NULL, "Right Out 1"}, + + {"Right Out 2", NULL, "Right Mixer"}, + {"ROUT2", NULL, "Right Out 2"}, + + {"Mono Mixer", "Left Playback Switch", "Left DAC"}, + {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Mono Mixer", "Right Playback Switch", "Right DAC"}, + {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Mono Out 1", NULL, "Mono Mixer"}, + {"MONO1", NULL, "Mono Out 1"}, + + {"Out3 Mux", "VREF", "VREF"}, + {"Out3 Mux", "ROUT1 + Vol", "ROUT1"}, + {"Out3 Mux", "ROUT1", "Right Mixer"}, + {"Out3 Mux", "MonoOut", "MONO1"}, + {"Out 3", NULL, "Out3 Mux"}, + {"OUT3", NULL, "Out 3"}, + + {"Left Line Mux", "Line 1", "LINPUT1"}, + {"Left Line Mux", "Line 2", "LINPUT2"}, + {"Left Line Mux", "Line 3", "LINPUT3"}, + {"Left Line Mux", "PGA", "Left PGA Mux"}, + {"Left Line Mux", "Differential", "Differential Mux"}, + + {"Right Line Mux", "Line 1", "RINPUT1"}, + {"Right Line Mux", "Line 2", "RINPUT2"}, + {"Right Line Mux", "Line 3", "RINPUT3"}, + /* {"Right Line Mux", "Mic", "MIC"}, */ + {"Right Line Mux", "PGA", "Right PGA Mux"}, + {"Right Line Mux", "Differential", "Differential Mux"}, + + {"Left PGA Mux", "Line 1", "LINPUT1"}, + {"Left PGA Mux", "Line 2", "LINPUT2"}, + {"Left PGA Mux", "Line 3", "LINPUT3"}, + {"Left PGA Mux", "Differential", "Differential Mux"}, + + {"Right PGA Mux", "Line 1", "RINPUT1"}, + {"Right PGA Mux", "Line 2", "RINPUT2"}, + {"Right PGA Mux", "Line 3", "RINPUT3"}, + {"Right PGA Mux", "Differential", "Differential Mux"}, + + {"Differential Mux", "Line 1", "LINPUT1"}, + {"Differential Mux", "Line 1", "RINPUT1"}, + {"Differential Mux", "Line 2", "LINPUT2"}, + {"Differential Mux", "Line 2", "RINPUT2"}, + + {"Left ADC Mux", "Stereo", "Left PGA Mux"}, + {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, + {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, + + {"Right ADC Mux", "Stereo", "Right PGA Mux"}, + {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, + {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, + + {"Left ADC", NULL, "Left ADC Mux"}, + {"Right ADC", NULL, "Right ADC Mux"}, +}; + + struct _coeff_div { u32 mclk; u32 rate; @@ -430,26 +852,90 @@ static int get_coeff(int mclk, int rate) return -EINVAL; } +/* The set of rates we can generate from the above for each SYSCLK */ +static const unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static const unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, 88200 +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static const unsigned int rates_18432[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list constraints_18432 = { + .count = ARRAY_SIZE(rates_18432), + .list = rates_18432, +}; + +static const unsigned int rates_169344[] = { + 8000, 11025, 22050, 44100, 88200 +}; + +static struct snd_pcm_hw_constraint_list constraints_169344 = { + .count = ARRAY_SIZE(rates_169344), + .list = rates_169344, +}; + +static const unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); switch (freq) { - case 11289600: - case 12000000: case 12288000: - case 16934400: + case 24576000: + wm8971->sysclk_constraints = &constraints_12288; + break; + case 11289600: + case 22579200: + wm8971->sysclk_constraints = &constraints_112896; + break; case 18432000: - wm8971->sysclk = freq; - return 0; + wm8971->sysclk_constraints = &constraints_18432; + break; + case 16934400: + case 33868800: + wm8971->sysclk_constraints = &constraints_169344; + break; + case 12000000: + case 24000000: + wm8971->sysclk_constraints = &constraints_12; + break; + default: + return -EINVAL; } - return -EINVAL; + + wm8971->sysclk = freq; + + return 0; } static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; @@ -507,8 +993,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -516,6 +1002,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; int coeff = get_coeff(wm8971->sysclk, params_rate(params)); + wm8971->playback_fs = params_rate(params); + /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -531,6 +1019,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, break; } + wm8973_set_deemph(codec); + /* set iface & srate */ snd_soc_write(codec, WM8971_IFACE, iface); if (coeff >= 0) @@ -553,23 +1043,31 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) } static int wm8971_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { - u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + static const u16 mask[][2] = { + [VARIANT_WM8971] = { 0x00c1, 0x0140 }, + [VARIANT_WM8973] = { 0x00c2, 0x0141 }, + }; + u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & + charge[wm8971->variant][0]; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | + mask[wm8971->variant][0]); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - snd_soc_cache_sync(codec); + regcache_sync(wm8971->regmap); /* mute dac and set vmid to 500k, enable VREF */ - snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | + mask[wm8971->variant][1]); break; case SND_SOC_BIAS_OFF: snd_soc_write(codec, WM8971_PWR1, 0x0001); @@ -580,8 +1078,9 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, } #define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -593,74 +1092,81 @@ static const struct snd_soc_dai_ops wm8971_dai_ops = { .set_sysclk = wm8971_set_dai_sysclk, }; -static struct snd_soc_dai_driver wm8971_dai = { - .name = "wm8971-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8971_RATES, - .formats = WM8971_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8971_RATES, - .formats = WM8971_FORMATS,}, - .ops = &wm8971_dai_ops, -}; - -static void wm8971_work(struct work_struct *work) -{ - struct snd_soc_dapm_context *dapm = - container_of(work, struct snd_soc_dapm_context, - delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; - wm8971_set_bias_level(codec, codec->dapm.bias_level); -} +static struct snd_soc_dai_driver wm8971_dai[] = { + { + .name = "wm8971-hifi-playback", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8971_RATES, + .formats = WM8971_FORMATS, + }, + .ops = &wm8971_dai_ops, + }, { + .name = "wm8971-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8971_RATES, + .formats = WM8971_FORMATS, + }, + .ops = &wm8971_dai_ops, + }, +}; static int wm8971_suspend(struct snd_soc_codec *codec) { + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8971->regmap); + return 0; } static int wm8971_resume(struct snd_soc_codec *codec) { + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 reg; wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8971 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; - snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); - } + if (codec->dapm.suspend_bias_level != SND_SOC_BIAS_ON) + return 0; + + reg = snd_soc_read(codec, WM8971_PWR1) & charge[wm8971->variant][0]; + snd_soc_write(codec, WM8971_PWR1, reg | charge[wm8971->variant][1]); + codec->dapm.bias_level = SND_SOC_BIAS_ON; return 0; } static int wm8971_probe(struct snd_soc_codec *codec) { + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); + const u32 *p; int ret = 0; u16 reg; - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; + codec->control_data = wm8971->regmap; wm8971_reset(codec); + if (codec->dev->of_node) { + p = of_get_property(codec->dev->of_node, "mclk-div", NULL); + if (p) + wm8971->mclk_div = be32_to_cpu(*p); + } + /* Master Clock Divide by 2 (0 = not div, 1 = div by 2) */ + if (wm8971->mclk_div) + snd_soc_update_bits(codec, WM8971_SRATE, 0x0040, 0x0040); /* charge output caps - set vmid to 5k for quick power up */ - reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; - snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & charge[wm8971->variant][0]; + snd_soc_write(codec, WM8971_PWR1, reg | charge[wm8971->variant][1]); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -681,24 +1187,47 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { - .probe = wm8971_probe, - .remove = wm8971_remove, - .suspend = wm8971_suspend, - .resume = wm8971_resume, - .set_bias_level = wm8971_set_bias_level, +struct regmap *wm8973_get_regmap(struct device *dev) +{ + struct wm8971_priv *priv = dev_get_drvdata(dev); + + return priv->regmap; +} - .controls = wm8971_snd_controls, - .num_controls = ARRAY_SIZE(wm8971_snd_controls), - .dapm_widgets = wm8971_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8971_dapm_widgets), - .dapm_routes = wm8971_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(wm8971_dapm_routes), +static struct snd_soc_codec_driver soc_codec_dev_wm8971[] = { + [VARIANT_WM8971] = { + .probe = wm8971_probe, + .remove = wm8971_remove, + .suspend = wm8971_suspend, + .resume = wm8971_resume, + .set_bias_level = wm8971_set_bias_level, + .get_regmap = wm8973_get_regmap, + + .controls = wm8971_snd_controls, + .num_controls = ARRAY_SIZE(wm8971_snd_controls), + .dapm_widgets = wm8971_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8971_dapm_widgets), + .dapm_routes = wm8971_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8971_dapm_routes), + }, + [VARIANT_WM8973] = { + .probe = wm8971_probe, + .remove = wm8971_remove, + .suspend = wm8971_suspend, + .resume = wm8971_resume, + .set_bias_level = wm8971_set_bias_level, + .get_regmap = wm8973_get_regmap, + + .controls = wm8973_snd_controls, + .num_controls = ARRAY_SIZE(wm8973_snd_controls), + .dapm_widgets = wm8973_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8973_dapm_widgets), + .dapm_routes = wm8973_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8973_dapm_routes), + } }; static const struct regmap_config wm8971_regmap = { @@ -715,7 +1244,6 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8971_priv *wm8971; - struct regmap *regmap; int ret; wm8971 = devm_kzalloc(&i2c->dev, sizeof(struct wm8971_priv), @@ -723,14 +1251,17 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, if (wm8971 == NULL) return -ENOMEM; - regmap = devm_regmap_init_i2c(i2c, &wm8971_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); + wm8971->variant = (int)id->driver_data; + + wm8971->regmap = devm_regmap_init_i2c(i2c, &wm8971_regmap); + if (IS_ERR(wm8971->regmap)) + return PTR_ERR(wm8971->regmap); i2c_set_clientdata(i2c, wm8971); ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8971, &wm8971_dai, 1); + &soc_codec_dev_wm8971[wm8971->variant], + wm8971_dai, ARRAY_SIZE(wm8971_dai)); return ret; } @@ -742,15 +1273,23 @@ static int wm8971_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8971_i2c_id[] = { - { "wm8971", 0 }, + { "wm8971", VARIANT_WM8971 }, + { "wm8973", VARIANT_WM8973 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); +static const struct of_device_id wm897x_dt_ids[] = { + { .compatible = "wlf,wm8971" }, + { .compatible = "wlf,wm8973" }, + { /* sentinel */ } +}; + static struct i2c_driver wm8971_i2c_driver = { .driver = { .name = "wm8971", .owner = THIS_MODULE, + .of_match_table = wm897x_dt_ids, }, .probe = wm8971_i2c_probe, .remove = wm8971_i2c_remove, @@ -762,3 +1301,5 @@ module_i2c_driver(wm8971_i2c_driver); MODULE_DESCRIPTION("ASoC WM8971 driver"); MODULE_AUTHOR("Lab126"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("*wm8971*"); +MODULE_ALIAS("*wm8973*"); diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h index f31c38f..aeab04a 100644 --- a/sound/soc/codecs/wm8971.h +++ b/sound/soc/codecs/wm8971.h @@ -22,21 +22,22 @@ #define WM8971_ADCDAC 0x05 #define WM8971_IFACE 0x07 #define WM8971_SRATE 0x08 -#define WM8971_LDAC 0x0a -#define WM8971_RDAC 0x0b -#define WM8971_BASS 0x0c +#define WM8971_LDAC 0x0a +#define WM8971_RDAC 0x0b +#define WM8971_BASS 0x0c #define WM8971_TREBLE 0x0d #define WM8971_RESET 0x0f -#define WM8971_ALC1 0x11 -#define WM8971_ALC2 0x12 -#define WM8971_ALC3 0x13 +#define WM8973_3D 0x10 +#define WM8971_ALC1 0x11 +#define WM8971_ALC2 0x12 +#define WM8971_ALC3 0x13 #define WM8971_NGATE 0x14 -#define WM8971_LADC 0x15 -#define WM8971_RADC 0x16 +#define WM8971_LADC 0x15 +#define WM8971_RADC 0x16 #define WM8971_ADCTL1 0x17 #define WM8971_ADCTL2 0x18 -#define WM8971_PWR1 0x19 -#define WM8971_PWR2 0x1a +#define WM8971_PWR1 0x19 +#define WM8971_PWR2 0x1a #define WM8971_ADCTL3 0x1b #define WM8971_ADCIN 0x1f #define WM8971_LADCIN 0x20 -- 1.7.9.5

2 1

[alsa-devel] generic i2s driver problems on an Exynos 5250 A15 board
by Matt Flax 12 Aug '14

12 Aug '14

hi there, I am writing a very generic alsa soc i2s driver for a very generic i2s device. The platform is an Exynos 5250 SoC board. Having trouble was looking for some help ... I have tried to start this driver in two ways. 1] Using the module_init/module_exit method and 2] Using the module_platform_driver approach. I have implemented the following : soc/codecs/implant.c soc/samsung/implant.c In both cases I have : static struct snd_soc_ops implant_ops = { .hw_params = implant_hw_params, }; static struct snd_soc_dai_link implant_dai = { .name = "implant", .stream_name = "IMPLANT", .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "implant-pcm", .platform_name = "samsung-i2s.0", .codec_name = "implant.0-001a", .ignore_suspend = 1, .ops = &implant_ops, }; I have setup the device tree as expected for sound@3830000 sound@3830000 { samsung,codec-type = "implant"; }; 1] Approach using module_init/module_exit static struct snd_soc_card snd_soc_machine_implant = { .name = "Implant", .owner = THIS_MODULE, .dai_link = &implant_dai, .num_links = 1, .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), }; Here is the output of the kernel : [ 1.802645] soc-audio soc-audio: ASoC: machine Implant should use snd_soc_register_card() [ 1.802656] soc-audio soc-audio: ASoC: CPU DAI samsung-i2s.0 not registered [ 1.802679] platform soc-audio: Driver soc-audio requests probe deferral [ 2.271140] soc-audio: probe of soc-audio failed with error -22 Not sure why it fails. The only thing I know is that my driver exits the init function returning 0 (module_init(implant_init)). This appears to be correct. 2] Approach using module_platform_driver When I switch to using snd_soc_register_card with : module_platform_driver(implant_driver); static struct platform_driver implant_driver = { .driver = { .name = "implant", .owner = THIS_MODULE, }, .probe = implant_probe, .remove = implant_remove, }; I get the following from dmesg : [ 1.802430] Error: Driver 'implant' is already registered, aborting... However I don't see any of my printk statements from dmesg ... Matt

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add merrifield machine driver
by Subhransu S. Prusty 12 Aug '14

12 Aug '14

From: Vinod Koul <vinod.koul(a)intel.com> This patch adds machine driver support for merrifield platform. Patch has dependency on merrifield DSP driver. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- Changes in v2: Removed get_codec/find_codec, instead use rtd index Using dai_fmt field of the dai_link This patch has dependency on "add mrfld DSP support". sound/soc/intel/Kconfig | 16 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/merr_dpcm_wm8958.c | 522 +++++++++++++++++++++++++++++++++++++ 3 files changed, 540 insertions(+) create mode 100644 sound/soc/intel/merr_dpcm_wm8958.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c6fe1e498a02..5f6345a6d77e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,19 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_MRFLD_WM8958_MACH + tristate "SOC Machine Audio driver for Intel Merrifield MID platform" + depends on X86 + select SND_SOC_WM8994 + select MFD_CORE + select MFD_WM8994 + select REGULATOR_WM8994 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + default n + help + This adds support for ASoC machine driver for Intel(R) MID Merrifield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9480b5d40e02..05da1093c777 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -25,7 +25,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-merr-dpcm-wm8958-objs := merr_dpcm_wm8958.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_MRFLD_WM8958_MACH) += snd-merr-dpcm-wm8958.o diff --git a/sound/soc/intel/merr_dpcm_wm8958.c b/sound/soc/intel/merr_dpcm_wm8958.c new file mode 100644 index 000000000000..7ef44a3ee608 --- /dev/null +++ b/sound/soc/intel/merr_dpcm_wm8958.c @@ -0,0 +1,522 @@ +/* + * merr_dpcm_wm8958.c - ASoc DPCM Machine driver for Intel Merrfield MID platform + * + * Copyright (C) 2013-14 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/async.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/mfd/wm8994/pdata.h> +#include "../codecs/wm8994.h" +#include "sst-atom-controls.h" + +/* Codec PLL output clk rate */ +#define CODEC_SYSCLK_RATE 24576000 +/* Input clock to codec at MCLK1 PIN */ +#define CODEC_IN_MCLK1_RATE 19200000 +/* Input clock to codec at MCLK2 PIN */ +#define CODEC_IN_MCLK2_RATE 32768 +/* define to select between MCLK1 and MCLK2 input to codec as its clock */ +#define CODEC_IN_MCLK1 1 +#define CODEC_IN_MCLK2 2 +#define CODEC_BE 1 + +struct mrfld_8958_mc_private { + struct snd_soc_jack jack; + int jack_retry; +}; + +/* + * Function to switch the input clock for codec, When audio is in + * progress input clock to codec will be through MCLK1 which is 19.2MHz + * while in off state input clock to codec will be through 32KHz through + * MCLK2 + * card : Sound card structure + * src : Input clock source to codec + */ +static int mrfld_8958_set_codec_clk(struct snd_soc_card *card, int src) +{ + int ret; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + switch (src) { + case CODEC_IN_MCLK1: + /* Turn ON the PLL to generate required sysclk rate + * from MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, + WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + CODEC_IN_MCLK1_RATE, CODEC_SYSCLK_RATE); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + /* Switch to MCLK1 input */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, + CODEC_SYSCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set codec sysclk configuration %d\n", + ret); + return ret; + } + break; + case CODEC_IN_MCLK2: + /* Switch to MCLK2 */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch to MCLK2: %d", ret); + return ret; + } + /* Turn off PLL for MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop the FLL: %d", ret); + return ret; + } + break; + default: + return -EINVAL; + } + return 0; +} + +static int mrfld_wm8958_set_clk(struct snd_soc_dai *codec_dai) +{ + int ret = 0; + struct snd_soc_card *card = codec_dai->card; + + pr_err("setting snd_soc_dai_set_tdm_slot\n"); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + pr_err("can't set codec pcm format %d\n", ret); + return ret; + } + + /* + * FIXME: move this to SYS_CLOCK event handler when codec driver + * dependency is clean. + */ + /* Switch to 19.2MHz MCLK1 input clock for codec */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK1); + + return ret; +} + +static int mrfld_8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + if (!strcmp(codec_dai->name, "wm8994-aif1")) + return mrfld_wm8958_set_clk(codec_dai); + return 0; +} + +static const struct snd_soc_pcm_stream mrfld_wm8958_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int merr_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + pr_debug("Invoked %s for dailink %s\n", __func__, rtd->dai_link->name); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int mrfld_8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + + ret = mrfld_wm8958_set_clk(aif1_dai); + break; + default: + break; + } + pr_debug("%s card(%s)->bias_level %u\n", __func__, card->name, + card->dapm.bias_level); + return ret; +} + +static int mrfld_8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + /* + * We are in standby down so */ + /* Switch to 32KHz MCLK2 input clock for codec + */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK2); + break; + default: + break; + } + if (&card->dapm == dapm) + card->dapm.bias_level = level; + pr_debug("%s card(%s)->bias_level %u\n", __func__, card->name, + card->dapm.bias_level); + return ret; +} + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_route map[] = { + { "Headphones", NULL, "HPOUT1L" }, + { "Headphones", NULL, "HPOUT1R" }, + + /* + * Machine uses 2 DMICs, other configs may use more so change below + * accordingly + */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, + /*{ "DMIC3DAT", NULL, "DMIC" },*/ + /*{ "DMIC4DAT", NULL, "DMIC" },*/ + + /* + * MICBIAS2 is connected as Bias for AMIC so we link it + * here. Also AMIC wires up to IN1LP pin. + * DMIC is externally connected to 1.8V rail, so no link rqd. + */ + { "AMIC", NULL, "MICBIAS2" }, + { "IN1LP", NULL, "AMIC" }, + + /* dsp map link the SWM outs to codec AIF */ + { "AIF1 Playback", NULL, "ssp2 Tx"}, + { "ssp2 Tx", NULL, "codec_out0"}, + { "ssp2 Tx", NULL, "codec_out1"}, + { "codec_in0", NULL, "ssp2 Rx" }, + { "codec_in1", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture"}, + +}; + +static const struct wm8958_micd_rate micdet_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, +}; + +static int mrfld_8958_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + struct mrfld_8958_mc_private *ctx = snd_soc_card_get_drvdata(card); + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + struct snd_soc_codec *codec = aif1_dai->codec; + + if (!aif1_dai) + return -ENODEV; + + pr_debug("Entry %s\n", __func__); + + ret = snd_soc_dai_set_tdm_slot(aif1_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + pr_err("can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + /* Force enable VMID to avoid cold latency constraints */ + snd_soc_dapm_force_enable_pin(&card->dapm, "VMID"); + snd_soc_dapm_sync(&card->dapm); + + if (!codec) { + pr_err("%s: we didnt find the codec pointer!\n", __func__); + return 0; + } + + ctx->jack_retry = 0; + ret = snd_soc_jack_new(codec, "Intel MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1, + &ctx->jack); + if (ret) { + pr_err("jack creation failed\n"); + return ret; + } + + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_1, KEY_MEDIA); + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + wm8958_mic_detect(codec, &ctx->jack, NULL, NULL, NULL, NULL); + + /* + * Micbias1 is always off, so for pm optimizations make sure the micbias1 + * discharge bit is set to floating to avoid discharge in disable state + */ + snd_soc_update_bits(codec, WM8958_MICBIAS1, WM8958_MICB1_DISCH, 0); + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; +static int mrfld_8958_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops mrfld_8958_ops = { + .startup = mrfld_8958_startup, +}; + +static struct snd_soc_ops mrfld_8958_be_ssp2_ops = { + .hw_params = mrfld_8958_hw_params, +}; + +struct snd_soc_dai_link mrfld_8958_msic_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Merrifield Audio Port", + .stream_name = "Merrifield Audio", + .cpu_dai_name = "Headset-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-platform", + .init = mrfld_8958_init, + .ignore_suspend = 1, + .dynamic = 1, + .ops = &mrfld_8958_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Merrifield Compress Port", + .stream_name = "Merrifield Compress", + .platform_name = "sst-platform", + .cpu_dai_name = "Compress-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .init = mrfld_8958_init, + }, + + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-platform", + .no_pcm = 1, + .codec_dai_name = "wm8994-aif1", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = merr_codec_fixup, + .ignore_suspend = 1, + .ops = &mrfld_8958_be_ssp2_ops, + }, +}; + +#ifdef CONFIG_PM_SLEEP +static int snd_mrfld_8958_prepare(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_codec *codec = card->rtd[CODEC_BE].codec; + struct snd_soc_dapm_context *dapm; + + pr_debug("In %s\n", __func__); + + pr_debug("found codec %s\n", codec->component.name); + dapm = &codec->dapm; + + snd_soc_dapm_disable_pin(dapm, "VMID"); + snd_soc_dapm_sync(dapm); + + snd_soc_suspend(dev); + return 0; +} + +static void snd_mrfld_8958_complete(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_codec *codec = card->rtd[CODEC_BE].codec; + struct snd_soc_dapm_context *dapm; + + pr_debug("In %s\n", __func__); + + pr_debug("found codec %s\n", codec->component.name); + dapm = &codec->dapm; + + snd_soc_dapm_force_enable_pin(dapm, "VMID"); + snd_soc_dapm_sync(dapm); + + snd_soc_resume(dev); +} + +static int snd_mrfld_8958_poweroff(struct device *dev) +{ + pr_debug("In %s\n", __func__); + snd_soc_poweroff(dev); + return 0; +} +#else +#define snd_mrfld_8958_prepare NULL +#define snd_mrfld_8958_complete NULL +#define snd_mrfld_8958_poweroff NULL +#endif + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mrfld = { + .name = "wm8958-audio", + .dai_link = mrfld_8958_msic_dailink, + .num_links = ARRAY_SIZE(mrfld_8958_msic_dailink), + .set_bias_level = mrfld_8958_set_bias_level, + .set_bias_level_post = mrfld_8958_set_bias_level_post, + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = map, + .num_dapm_routes = ARRAY_SIZE(map), + .fully_routed = true, +}; + +static int snd_mrfld_8958_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct mrfld_8958_mc_private *drv; + + pr_debug("Entry %s\n", __func__); + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (!drv) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + + /* register the soc card */ + snd_soc_card_mrfld.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_mrfld, drv); + ret_val = snd_soc_register_card(&snd_soc_card_mrfld); + if (ret_val) { + pr_err("snd_soc_register_card failed %d\n", ret_val); + goto unalloc; + } + platform_set_drvdata(pdev, &snd_soc_card_mrfld); + pr_info("%s successful\n", __func__); + return ret_val; + +unalloc: + return ret_val; +} + +static int snd_mrfld_8958_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + pr_debug("In %s\n", __func__); + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +const struct dev_pm_ops snd_mrfld_8958_mc_pm_ops = { + .prepare = snd_mrfld_8958_prepare, + .complete = snd_mrfld_8958_complete, + .poweroff = snd_mrfld_8958_poweroff, +}; + +static struct platform_driver snd_mrfld_8958_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "mrfld_wm8958", + .pm = &snd_mrfld_8958_mc_pm_ops, + }, + .probe = snd_mrfld_8958_mc_probe, + .remove = snd_mrfld_8958_mc_remove, +}; + +module_platform_driver(snd_mrfld_8958_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Merrifield MID Machine driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mrfld_wm8958"); -- 1.9.0

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[alsa-devel] [PATCH] ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double()
by Geert Uytterhoeven 11 Aug '14

11 Aug '14

If soc_dapm_read() fails, reg_val will be uninitialized, and bogus values will be written later: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double': sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized] unsigned int reg_val, val; ^ Return early on error to fix this. Introduced by commit ce0fc93ae56e2ba50ff8c220d69e4e860e889320 ("ASoC: Add DAPM support at the component level"). --- Is this correct? Please review. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/soc-dapm.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c6..177bd8639ef9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - int ret = 0; - if (e->reg != SND_SOC_NOPM) - ret = soc_dapm_read(dapm, e->reg, &reg_val); - else + if (e->reg != SND_SOC_NOPM) { + int ret = soc_dapm_read(dapm, e->reg, &reg_val); + if (ret) + return ret; + } else { reg_val = dapm_kcontrol_get_value(kcontrol); + } val = (reg_val >> e->shift_l) & e->mask; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); @@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: Intel: Baytrail: Fix stream restore after aborted suspend
by Jarkko Nikula 11 Aug '14

11 Aug '14

It was found that resuming active stream fails on Baytrail ADSP in case of aborted suspend or when testing only partial suspend. Reason for this is that in those cases ADSP doesn't reach reset even though resume path is expecting so and fails to restart stream. This revealed that suspend and resume phases do not go together: Operating state is not the same after returning certain resume phase than it was prior entering into corresponding suspend phase. Also we unload firmware in suspend_late before putting ADSP into reset in suspend_noirq when ADSP is actually already powered down. Fix these by first rearranging and simplifying suspend/resume: Put ADSP into reset in suspend_late and bring it back in resume_late. Now only these will take care of ADSP reset and reboot and SNDRV_PCM_TRIGGER_SUSPEND/SNDRV_PCM_TRIGGER_RESUME callbacks will take care of stream stop/restart during suspend and resume phases. Then modify stream resume so that full stream reallocation and restart is done only if suspend_late was reached. Otherwise resume it by using the IPC that resumes paused stream. Jarkko Nikula (3): ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_late ASoC: Intel: Wait Baytrail ADSP boot at resume_early stage ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset sound/soc/intel/sst-baytrail-ipc.c | 10 +-------- sound/soc/intel/sst-baytrail-ipc.h | 1 - sound/soc/intel/sst-baytrail-pcm.c | 43 +++++++++++++------------------------- 3 files changed, 16 insertions(+), 38 deletions(-) -- 2.0.1

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[alsa-devel] Choosing the sysclk in simple-card looks broken to me.
by jonsmirl＠gmail.com 11 Aug '14

11 Aug '14

The problem is in asoc_simple_card_sub_parse_of() in the else case.. } else { clk = of_clk_get(node, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } This is picking up the input clocks to the devices. On the cpu-dai you want to pick up the ouput mclk (my first input clock is an 80Mhz bus clock). My IIS output clock is: clock-output-names = "mclk0"; Setting clocks="" doesn't work either, it also picks up the input clock. if (of_property_read_bool(np, "clocks")) { clk = of_clk_get(np, 0); if (IS_ERR(clk)) { ret = PTR_ERR(clk); return ret; When the codec-dai bit runs it finds the input clock on the codec, which is really my cpu-dai's output clock and it gets the correct value. But then simple goes and tries to set both system clocks - my 80Mhz bus clock and the 22.5 true sysclk. And my cpu-dai barfs when it is passed a 80Mhz sysclk. How to fix? Is anyone using this feature? This needs to be fixed to pick up output clocks, not input ones. -- Jon Smirl jonsmirl(a)gmail.com

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