From: Graeme Gregory graeme@openmoko.com

Signed-off-by: Graeme Gregory graeme@openmoko.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/s3c24xx/s3c24xx-i2s.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 38 insertions(+), 0 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index cd89c41..fee7925 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -75,6 +75,10 @@ static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { struct s3c24xx_i2s_info { void __iomem *regs; struct clk *iis_clk; + u32 iiscon; + u32 iismod; + u32 iisfcon; + u32 iispsr; }; static struct s3c24xx_i2s_info s3c24xx_i2s;

@@ -405,6 +409,38 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) return 0; }

+#ifdef CONFIG_PM +int s3c24xx_i2s_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); + + clk_disable(s3c24xx_i2s.iis_clk); + + return 0; +} + +int s3c24xx_i2s_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + clk_enable(s3c24xx_i2s.iis_clk); + + writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); + + return 0; +} +#else +#define s3c24xx_i2s_suspend NULL +#define s3c24xx_i2s_resume NULL +#endif + + #define S3C24XX_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -415,6 +451,8 @@ struct snd_soc_cpu_dai s3c24xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = s3c24xx_i2s_probe, + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, .playback = { .channels_min = 2, .channels_max = 2,

From: Liam Girdwood liam@llanelli.wolfsonmicro.main

Added a device level dapm event so that both the machine and codec are informed when dapm events occur.

Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 23 +++++++++++------------ sound/soc/soc-dapm.c | 23 +++++++++++++++++++++++ 4 files changed, 38 insertions(+), 12 deletions(-)

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2b1ae8e..db02159 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -199,6 +199,7 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event);

/* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index f47ef1f..000f3e7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -426,6 +426,9 @@ struct snd_soc_machine { int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev);

+ /* callbacks */ + int (*dapm_event)(struct snd_soc_machine *, int event); + /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e6a67b5..aaf141c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -289,15 +289,16 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) {

codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, + snd_soc_dapm_stream_event(codec, + codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP);

/* power down the codec power domain if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D3hot); } } } @@ -353,12 +354,12 @@ static int soc_codec_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); + codec_dai->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP);

- if (codec->active == 0 && codec_dai->pop_wait == 0){ - if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); - } + if (codec->active == 0 && codec_dai->pop_wait == 0) + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D3hot); }

mutex_unlock(&pcm_mutex); @@ -433,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* no delayed work - do we need to power up codec */ if (codec->dapm_state != SNDRV_CTL_POWER_D0) {

- if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D1); + snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1);

if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -445,8 +445,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START);

- if (codec->dapm_event) - codec->dapm_event(codec, SNDRV_CTL_POWER_D0); + snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); if (codec_dai->dai_ops.digital_mute) codec_dai->dai_ops.digital_mute(codec_dai, 0);

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 29a546f..0a9d192 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1280,6 +1280,29 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);

/** + * snd_soc_dapm_device_event - send a device event to the dapm core + * @socdev: audio device + * @event: device event + * + * Sends a device event to the dapm core. The core then makes any + * necessary machine or codec power changes.. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +{ + struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_machine *machine = socdev->machine; + + if (machine->dapm_event) + machine->dapm_event(machine, event); + if (codec->dapm_event) + codec->dapm_event(codec, event); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); + +/** * snd_soc_dapm_set_endpoint - set audio endpoint status * @codec: audio codec * @endpoint: audio signal endpoint (or start point)

From: Liam Girdwood liam@localhost.localdomain

--- sound/soc/soc-core.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 265bd55..f3e1b74 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -288,6 +288,14 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) {

+ /* power down the codec to D1 if no longer active */ + if (codec->active == 0) { + dbg("pop wq D1 %s %s\n", codec->name, + codec_dai->playback.stream_name); + snd_soc_dapm_device_event(socdev, + SNDRV_CTL_POWER_D1); + } + codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Milan plzik milan.plzik@gmail.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com --- sound/soc/soc-dapm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d5d3c17..e8d7fae 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -692,7 +692,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; }

-/* test and update the power status of a mixer widget */ +/* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int reg, int val_mask, int val, int invert) @@ -700,7 +700,8 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_soc_dapm_path *path; int found = 0;

- if (widget->id != snd_soc_dapm_mixer) + if (widget->id != snd_soc_dapm_mixer && + widget->id != snd_soc_dapm_switch) return -ENODEV;

if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))

From: Liam Girdwood lg@opensource.wolfsonmicro.com

Signed-off-by: Laim Girdwood lg@opensource.wolfsonmicro.com --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-dapm.c | 40 +++++++++++++++++++++++++++------------- 2 files changed, 28 insertions(+), 14 deletions(-)

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 335d73c..0df89bb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -289,7 +289,7 @@ struct snd_soc_dapm_widget {

/* external events */ unsigned short event_flags; /* flags to specify event types */ - int (*event)(struct snd_soc_dapm_widget*, int); + int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);

/* kcontrols that relate to this widget */ int num_kcontrols; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4a5bd85..a4ce882 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -524,11 +524,13 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue;

if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); if (ret < 0) return ret; } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); if (ret < 0) return ret; } @@ -539,11 +541,13 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue;

if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, SND_SOC_DAPM_POST_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); if (ret < 0) return ret; } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, SND_SOC_DAPM_POST_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); if (ret < 0) return ret; } @@ -567,26 +571,30 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); if (ret < 0) return ret; } dapm_update_bits(w); if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, SND_SOC_DAPM_POST_PMU); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); if (ret < 0) return ret; } } else { /* power down event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, SND_SOC_DAPM_PRE_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); if (ret < 0) return ret; } dapm_update_bits(w); if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, SND_SOC_DAPM_POST_PMD); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); if (ret < 0) return ret; } @@ -1096,13 +1104,17 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); - if (ret < 0) + ret = widget->event(widget, kcontrol, + SND_SOC_DAPM_PRE_REG); + if (ret < 0) { + ret = 1; goto out; + } } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, SND_SOC_DAPM_POST_REG); + ret = widget->event(widget, kcontrol, + SND_SOC_DAPM_POST_REG); } else ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);

@@ -1177,13 +1189,15 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, dapm_mux_update_power(widget, kcontrol, mask, mux, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); if (ret < 0) goto out; } ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, SND_SOC_DAPM_POST_REG); + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); } else ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Marc Mulcahy marc@levelstar.com Signed-off-by: Graeme Gregory gg@opensource.wolfsonmicro.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9713.c | 1289 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9713.h | 53 ++ 4 files changed, 1348 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/wm9713.c create mode 100644 sound/soc/codecs/wm9713.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7824880..e0675ef 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,10 @@ config SND_SOC_WM9712 tristate depends on SND_SOC

+config SND_SOC_WM9713 + tristate + depends on SND_SOC + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate @@ -36,4 +40,3 @@ config SND_SOC_CS4270_HWMUTE config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 - diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ad78e3..f64eb07 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o +snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o

obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o @@ -10,4 +11,5 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c new file mode 100644 index 0000000..7eaa034 --- /dev/null +++ b/sound/soc/codecs/wm9713.c @@ -0,0 +1,1289 @@ +/* + * wm9713.c -- ALSA Soc WM9713 codec support + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 4th Feb 2006 Initial version. + * + * Features:- + * + * o Support for AC97 Codec, Voice DAC and Aux DAC + * o Support for DAPM + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "wm9713.h" + +#define WM9713_VERSION "0.15" + +struct wm9713_priv { + u32 pll_in; /* PLL input frequency */ + u32 pll_out; /* PLL output frequency */ +}; + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg); +static int ac97_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val); + +/* + * WM9713 register cache + * Reg 0x3c bit 15 is used by touch driver. + */ +static const u16 wm9713_reg[] = { + 0x6174, 0x8080, 0x8080, 0x8080, + 0xc880, 0xe808, 0xe808, 0x0808, + 0x00da, 0x8000, 0xd600, 0xaaa0, + 0xaaa0, 0xaaa0, 0x0000, 0x0000, + 0x0f0f, 0x0040, 0x0000, 0x7f00, + 0x0405, 0x0410, 0xbb80, 0xbb80, + 0x0000, 0xbb80, 0x0000, 0x4523, + 0x0000, 0x2000, 0x7eff, 0xffff, + 0x0000, 0x0000, 0x0080, 0x0000, + 0x0000, 0x0000, 0xfffe, 0xffff, + 0x0000, 0x0000, 0x0000, 0xfffe, + 0x4000, 0x0000, 0x0000, 0x0000, + 0xb032, 0x3e00, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0006, + 0x0001, 0x0000, 0x574d, 0x4c13, + 0x0000, 0x0000, 0x0000 +}; + +/* virtual HP mixers regs */ +#define HPL_MIXER 0x80 +#define HPR_MIXER 0x82 +#define MICB_MUX 0x82 + +static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; +static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; +static const char *wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", + "Mono Out", "Zh"}; +static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", + "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_spk_pga[] = + {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", + "Speaker Vmid", "Inv Vmid"}; +static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", + "Headphone Vmid"}; +static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; +static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; +static const char *wm9713_dac_inv[] = + {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", + "Headphone Mono", "NC", "Vmid"}; +static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; +static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; +static const char *wm9713_micb_select[] = {"MPB", "MPA"}; + +static const struct soc_enum wm9713_enum[] = { +SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ +SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ +SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ +SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ +SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ +}; + +static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { +SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), +SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), +SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), +SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), +SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), + +SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), + +SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), +SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), +SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), +SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), + +SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), +SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), + +SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +SOC_ENUM("ALC Function", wm9713_enum[6]), +SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), +SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), +SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +SOC_ENUM("ALC NG Type", wm9713_enum[17]), +SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), + +SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), +SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), + +SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), +SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), +SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), + +SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), +SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), +SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), + +SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), +SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), + +SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), + +SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), +SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), + +SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), +SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), +SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), + +SOC_ENUM("Bass Control", wm9713_enum[16]), +SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), +SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), +SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), +SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), +SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), + +SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), +SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), +SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), +}; + +/* add non dapm controls */ +static int wm9713_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm9713_snd_ac97_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* We have to create a fake left and right HP mixers because + * the codec only has a single control that is shared by both channels. + * This makes it impossible to determine the audio path using the current + * register map, thus we add a new (virtual) register to help determine the + * audio route within the device. + */ +static int mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 l, r, beep, tone, phone, rec, pcm, aux; + + l = ac97_read(w->codec, HPL_MIXER); + r = ac97_read(w->codec, HPR_MIXER); + beep = ac97_read(w->codec, AC97_PC_BEEP); + tone = ac97_read(w->codec, AC97_MASTER_TONE); + phone = ac97_read(w->codec, AC97_PHONE); + rec = ac97_read(w->codec, AC97_REC_SEL); + pcm = ac97_read(w->codec, AC97_PCM); + aux = ac97_read(w->codec, AC97_AUX); + + if (event & SND_SOC_DAPM_PRE_REG) + return 0; + if (l & 0x1 || r & 0x1) + ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); + else + ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + + if (l & 0x2 || r & 0x2) + ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); + else + ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); + + if (l & 0x4 || r & 0x4) + ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); + else + ac97_write(w->codec, AC97_PHONE, phone | 0x8000); + + if (l & 0x8 || r & 0x8) + ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); + else + ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); + + if (l & 0x10 || r & 0x10) + ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); + else + ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + + if (l & 0x20 || r & 0x20) + ac97_write(w->codec, AC97_AUX, aux & 0x7fff); + else + ac97_write(w->codec, AC97_AUX, aux | 0x8000); + + return 0; +} + +/* Left Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), +}; + +/* Right Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), +}; + +/* headphone capture mux */ +static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[1]); + +/* headphone mic mux */ +static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[0]); + +/* Speaker Mixer */ +static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), +SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), +SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), +}; + +/* mono mic mux */ +static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[2]); + +/* mono output mux */ +static const struct snd_kcontrol_new wm9713_mono_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[7]); + +/* speaker left output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[8]); + +/* speaker right output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[9]); + +/* headphone left output mux */ +static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[10]); + +/* headphone right output mux */ +static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[11]); + +/* Out3 mux */ +static const struct snd_kcontrol_new wm9713_out3_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[12]); + +/* Out4 mux */ +static const struct snd_kcontrol_new wm9713_out4_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[13]); + +/* DAC inv mux 1 */ +static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[14]); + +/* DAC inv mux 2 */ +static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[15]); + +/* Capture source left */ +static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[3]); + +/* Capture source right */ +static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[4]); + +/* mic source */ +static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[18]); + +/* mic source B virtual control */ +static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[19]); + +static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { +SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_rec_mux_controls), +SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_mic_mux_controls), +SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mic_mux_controls), +SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mux_controls), +SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkl_mux_controls), +SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkr_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out3_mux_controls), +SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out4_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv1_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv2_mux_controls), +SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcl_mux_controls), +SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcr_mux_controls), +SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, + &wm9713_mic_sel_mux_controls), +SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, + &wm9713_micb_sel_mux_controls), +SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, + &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, + &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, + &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, + &wm9713_speaker_mixer_controls[0], + ARRAY_SIZE(wm9713_speaker_mixer_controls)), +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), +SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), +SND_SOC_DAPM_OUTPUT("MONO"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKL"), +SND_SOC_DAPM_OUTPUT("SPKR"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_INPUT("LINEL"), +SND_SOC_DAPM_INPUT("LINER"), +SND_SOC_DAPM_INPUT("MONOIN"), +SND_SOC_DAPM_INPUT("PCBEEP"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2A"), +SND_SOC_DAPM_INPUT("MIC2B"), +SND_SOC_DAPM_VMID("VMID"), +}; + +static const char *audio_map[][3] = { + /* left HP mixer */ + {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, + {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Left HP Mixer", NULL, "Capture Headphone Mux"}, + + /* right HP mixer */ + {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, + {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Right HP Mixer", NULL, "Capture Headphone Mux"}, + + /* virtual mixer - mixes left & right channels for spk and mono */ + {"AC97 Mixer", NULL, "Left DAC"}, + {"AC97 Mixer", NULL, "Right DAC"}, + {"Line Mixer", NULL, "Right Line In"}, + {"Line Mixer", NULL, "Left Line In"}, + {"HP Mixer", NULL, "Left HP Mixer"}, + {"HP Mixer", NULL, "Right HP Mixer"}, + {"Capture Mixer", NULL, "Left Capture Source"}, + {"Capture Mixer", NULL, "Right Capture Source"}, + + /* speaker mixer */ + {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, + + /* mono mixer */ + {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", NULL, "Capture Mono Mux"}, + + /* DAC inv mux 1 */ + {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, + + /* DAC inv mux 2 */ + {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, + + /* headphone left mux */ + {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, + + /* headphone right mux */ + {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, + + /* speaker left mux */ + {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, + {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* speaker right mux */ + {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, + {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, + + /* mono mux */ + {"Mono Out Mux", "Mono", "Mono Mixer"}, + {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* out 3 mux */ + {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, + + /* out 4 mux */ + {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Out Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Out Mux"}, + {"OUT3", NULL, "Out 3"}, + {"Out 3", NULL, "Out 3 Mux"}, + {"OUT4", NULL, "Out 4"}, + {"Out 4", NULL, "Out 4 Mux"}, + {"SPKL", NULL, "Left Speaker"}, + {"Left Speaker", NULL, "Left Speaker Out Mux"}, + {"SPKR", NULL, "Right Speaker"}, + {"Right Speaker", NULL, "Right Speaker Out Mux"}, + {"MONO", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono Out Mux"}, + + /* input pga */ + {"Left Line In", NULL, "LINEL"}, + {"Right Line In", NULL, "LINER"}, + {"Mono In", NULL, "MONOIN"}, + {"Mic A PGA", NULL, "Mic A Pre Amp"}, + {"Mic B PGA", NULL, "Mic B Pre Amp"}, + + /* left capture select */ + {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Left Capture Source", "Line", "LINEL"}, + {"Left Capture Source", "Mono In", "MONOIN"}, + {"Left Capture Source", "Headphone", "Left HP Mixer"}, + {"Left Capture Source", "Speaker", "Speaker Mixer"}, + {"Left Capture Source", "Mono Out", "Mono Mixer"}, + + /* right capture select */ + {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Right Capture Source", "Line", "LINER"}, + {"Right Capture Source", "Mono In", "MONOIN"}, + {"Right Capture Source", "Headphone", "Right HP Mixer"}, + {"Right Capture Source", "Speaker", "Speaker Mixer"}, + {"Right Capture Source", "Mono Out", "Mono Mixer"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Source"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Source"}, + + /* mic */ + {"Mic A Pre Amp", NULL, "Mic A Source"}, + {"Mic A Source", "Mic 1", "MIC1"}, + {"Mic A Source", "Mic 2 A", "MIC2A"}, + {"Mic A Source", "Mic 2 B", "Mic B Source"}, + {"Mic B Pre Amp", "MPB", "Mic B Source"}, + {"Mic B Source", NULL, "MIC2B"}, + + /* headphone capture */ + {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, + {"Capture Headphone Mux", "Left", "Left Capture Source"}, + {"Capture Headphone Mux", "Right", "Right Capture Source"}, + + /* mono capture */ + {"Capture Mono Mux", "Stereo", "Capture Mixer"}, + {"Capture Mono Mux", "Left", "Left Capture Source"}, + {"Capture Mono Mux", "Right", "Right Capture Source"}, + + {NULL, NULL, NULL}, +}; + +static int wm9713_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + + /* set up audio path audio_mapnects */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || + reg == AC97_CD) + return soc_ac97_ops.read(codec->ac97, reg); + else { + reg = reg >> 1; + + if (reg > (ARRAY_SIZE(wm9713_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg <= (ARRAY_SIZE(wm9713_reg))) + cache[reg] = val; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 divsel:1; + u32 divctl:1; + u32 lf:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the PLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 22) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + /* The the PLL output is always 98.304MHz. */ + target = 98304000; + + /* If the input frequency is over 14.4MHz then scale it down. */ + if (source > 14400000) { + source >>= 1; + pll_div->divsel = 1; + + if (source > 14400000) { + source >>= 1; + pll_div->divctl = 1; + } else + pll_div->divctl = 0; + + } else { + pll_div->divsel = 0; + pll_div->divctl = 0; + } + + /* Low frequency sources require an additional divide in the + * loop. + */ + if (source < 8192000) { + pll_div->lf = 1; + target >>= 2; + } else + pll_div->lf = 0; + + Ndiv = target / source; + if ((Ndiv < 5) || (Ndiv > 12)) + printk(KERN_WARNING + "WM9713 PLL N value %d out of recommended range!\n", + Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +/** + * Please note that changing the PLL input frequency may require + * resynchronisation with the AC97 controller. + */ +static int wm9713_set_pll(struct snd_soc_codec *codec, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct wm9713_priv *wm9713 = codec->private_data; + u16 reg, reg2; + struct _pll_div pll_div; + + /* turn PLL off ? */ + if (freq_in == 0 || freq_out == 0) { + /* disable PLL power and select ext source */ + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + wm9713->pll_out = 0; + return 0; + } + + pll_factors(&pll_div, freq_in); + + if (pll_div.k == 0) { + reg = (pll_div.n << 12) | (pll_div.lf << 11) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } else { + /* write the fractional k to the reg 0x46 pages */ + reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + + /* K [21:20] */ + reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [19:16] */ + reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [15:12] */ + reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [11:8] */ + reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [7:4] */ + reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } + + /* turn PLL on and select as source */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + wm9713->pll_out = freq_out; + wm9713->pll_in = freq_in; + + /* wait 10ms AC97 link frames for the link to stabilise */ + schedule_timeout_interruptible(msecs_to_jiffies(10)); + return 0; +} + +static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + return wm9713_set_pll(codec, pll_id, freq_in, freq_out); +} + +/* + * Tristate the PCM DAI lines, tristate can be disabled by calling + * wm9713_set_dai_fmt() + */ +static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, + int tristate) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; + + if (tristate) + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + + return 0; +} + +/* + * Configure WM9713 clock dividers. + * Voice DAC needs 256 FS + */ +static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM9713_PCMCLK_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKA_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKB_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_HIFI_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_PCMBCLK_DIV: + reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + break; + case WM9713_PCMCLK_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + break; + case WM9713_HIFI_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + break; + default: + return -EINVAL; + } + + return 0; +}; + +static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 reg = 0x8000; + + /* clock masters */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= 0x4000; + gpio |= 0x0010; + break; + case SND_SOC_DAIFMT_CBM_CFS: + reg |= 0x6000; + gpio |= 0x0018; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= 0x0200; + gpio |= 0x001a; + break; + case SND_SOC_DAIFMT_CBS_CFM: + gpio |= 0x0012; + break; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + reg |= 0x00c0; + break; + case SND_SOC_DAIFMT_IB_NF: + reg |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= 0x0040; + break; + } + + /* DAI format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + reg |= 0x0043; + break; + } + + ac97_write(codec, AC97_GPIO_CFG, gpio); + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static int wm9713_pcm_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 0x000c; + break; + } + + /* enable PCM interface in master mode */ + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 status; + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; + ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_EXTENDED_MID, status); +} + +static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +static int ac97_aux_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 vra, xsle; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + xsle = ac97_read(codec, AC97_PCI_SID); + ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + + return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); +} + +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_FORMATS \ + (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_codec_dai wm9713_dai[] = { +{ + .name = "AC97 HiFi", + .type = SND_SOC_DAI_AC97_BUS, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_hifi_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_aux_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "WM9713 Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .ops = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(wm9713_dai); + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (!(ac97_read(codec, 0) & 0x8000)) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) & 0x8000) + return -EIO; + return 0; +} +EXPORT_SYMBOL_GPL(wm9713_reset); + +static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 reg; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* enable thermal shutdown */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* enable master bias and vmid */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* disable everything including AC link */ + ac97_write(codec, AC97_EXTENDED_MID, 0xffff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->dapm_state = event; + return 0; +} + +static int wm9713_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int wm9713_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm9713_priv *wm9713 = codec->private_data; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* do we need to re-start the PLL ? */ + if (wm9713->pll_out) + wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + + /* only synchronise the codec if warm reset failed */ + if (ret == 0) { + for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { + if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || + i == AC97_EXTENDED_MSTATUS || i > 0x66) + continue; + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + } + + if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + + return ret; +} + +static int wm9713_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0, reg; + + printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9713_reg); + codec->reg_cache_step = 2; + + codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL); + if (codec->private_data == NULL) { + ret = -ENOMEM; + goto priv_err; + } + + codec->name = "WM9713"; + codec->owner = THIS_MODULE; + codec->dai = wm9713_dai; + codec->num_dai = ARRAY_SIZE(wm9713_dai); + codec->write = ac97_write; + codec->read = ac97_read; + codec->dapm_event = wm9713_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + wm9713_reset(codec, 0); + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "AC97 link error\n"); + goto reset_err; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* unmute the adc - move to kcontrol */ + reg = ac97_read(codec, AC97_CD) & 0x7fff; + ac97_write(codec, AC97_CD, reg); + + wm9713_add_controls(codec); + wm9713_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + +pcm_err: + snd_soc_free_ac97_codec(codec); + +codec_err: + kfree(codec->private_data); + +priv_err: + kfree(codec->reg_cache); + +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9713_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec->dai); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9713 = { + .probe = wm9713_soc_probe, + .remove = wm9713_soc_remove, + .suspend = wm9713_soc_suspend, + .resume = wm9713_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); + +MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h new file mode 100644 index 0000000..d357b6c --- /dev/null +++ b/sound/soc/codecs/wm9713.h @@ -0,0 +1,53 @@ +/* + * wm9713.h -- WM9713 Soc Audio driver + */ + +#ifndef _WM9713_H +#define _WM9713_H + +/* clock inputs */ +#define WM9713_CLKA_PIN 0 +#define WM9713_CLKB_PIN 1 + +/* clock divider ID's */ +#define WM9713_PCMCLK_DIV 0 +#define WM9713_CLKA_MULT 1 +#define WM9713_CLKB_MULT 2 +#define WM9713_HIFI_DIV 3 +#define WM9713_PCMBCLK_DIV 4 +#define WM9713_PCMCLK_PLL_DIV 5 +#define WM9713_HIFI_PLL_DIV 6 + +/* Calculate the appropriate bit mask for the external PCM clock divider */ +#define WM9713_PCMDIV(x) ((x - 1) << 8) + +/* Calculate the appropriate bit mask for the external HiFi clock divider */ +#define WM9713_HIFIDIV(x) ((x - 1) << 12) + +/* MCLK clock mulitipliers */ +#define WM9713_CLKA_X1 (0 << 1) +#define WM9713_CLKA_X2 (1 << 1) +#define WM9713_CLKB_X1 (0 << 2) +#define WM9713_CLKB_X2 (1 << 2) + +/* MCLK clock MUX */ +#define WM9713_CLK_MUX_A (0 << 0) +#define WM9713_CLK_MUX_B (1 << 0) + +/* Voice DAI BCLK divider */ +#define WM9713_PCMBCLK_DIV_1 (0 << 9) +#define WM9713_PCMBCLK_DIV_2 (1 << 9) +#define WM9713_PCMBCLK_DIV_4 (2 << 9) +#define WM9713_PCMBCLK_DIV_8 (3 << 9) +#define WM9713_PCMBCLK_DIV_16 (4 << 9) + +#define WM9713_DAI_AC97_HIFI 0 +#define WM9713_DAI_AC97_AUX 1 +#define WM9713_DAI_PCM_VOICE 2 + +extern struct snd_soc_codec_device soc_codec_dev_wm9713; +extern struct snd_soc_codec_dai wm9713_dai[3]; + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm); + +#endif

From: Richard Purdie richard@openedhand.com

Signed-off-by: Richard Purdie richard@openedhand.com Signed-off-by: Milan Plzik milan.plzik@gmail.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com Cc: i2c@lm-sensors.org --- include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4535.c | 716 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4535.h | 46 +++ 5 files changed, 769 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4535.c create mode 100644 sound/soc/codecs/ak4535.h

diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index 88c8140..5ced329 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -121,6 +121,7 @@ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ +#define I2C_DRIVERID_AK4535 95 /* AK4525 audio codec */

#define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0675ef..8244107 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,10 @@ config SND_SOC_AC97_CODEC tristate depends on SND_SOC

+config SND_SOC_AK4535 + tristate + depends on SND_SOC + config SND_SOC_WM8731 tristate depends on SND_SOC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f64eb07..6859145 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ak4535-objs := ak4535.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -7,6 +8,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o

obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c new file mode 100644 index 0000000..0db47ef --- /dev/null +++ b/sound/soc/codecs/ak4535.c @@ -0,0 +1,716 @@ +/* + * ak4535.c -- AK4535 ALSA Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie richard@openedhand.com + * + * Based on wm8753.c by Liam Girdwood + * + * 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 <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4535.h" + +#define AUDIO_NAME "ak4535" +#define AK4535_VERSION "0.3" + +#define AK4535_DEBUG 0 +#if AK4535_DEBUG +#else +#define ak4535dbg_dump(s) if (0) {} +#endif + +struct snd_soc_codec_device soc_codec_dev_ak4535; + +/* codec private data */ +struct ak4535_priv { + unsigned int sysclk; +}; + +/* + * ak4535 register cache + */ +static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x0000, 0x0080, 0x0000, 0x0003, + 0x0002, 0x0000, 0x0011, 0x0001, + 0x0000, 0x0040, 0x0036, 0x0010, + 0x0000, 0x0000, 0x0057, 0x0000, +}; + +/* + * read ak4535 register cache + */ +static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4535_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4535 register cache + */ +static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the AK4535 register space + */ +static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4535 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4535_write_reg_cache (codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int ak4535_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r=0; + + for (i = 0; i < AK4535_CACHEREGNUM; i++) + r |= ak4535_write (codec, i, cache[i]); + + ak4535dbg_dump (codec); + + return r; +}; + +static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; +static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; +static const char *ak4535_mic_select[] = {"Internal", "External"}; + +static const struct soc_enum ak4535_enum[] = { + SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), + SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), + SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), + SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), + SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), +}; + +static const struct snd_kcontrol_new ak4535_snd_controls[] = { + SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), + SOC_ENUM("Mono 1 Output", ak4535_enum[1]), + SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), + SOC_ENUM("Headphone Output", ak4535_enum[2]), + SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), + SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), + SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), + SOC_ENUM("Mic Select", ak4535_enum[4]), + SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), + SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), + SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), + SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), + SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), + SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), + SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), + SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), + SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), +}; + +/* add non dapm controls */ +static int ak4535_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ak4535_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4535_enum[4]); + +/* HP L switch */ +static const struct snd_kcontrol_new ak4535_hpl_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); + +/* HP R switch */ +static const struct snd_kcontrol_new ak4535_hpr_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4535_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); + +/* Line out switch */ +static const struct snd_kcontrol_new ak4535_line_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); + +/* ak4535 dapm widgets */ +static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_stereo_mixer_controls[0], + ARRAY_SIZE(ak4535_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_mono1_mixer_controls[0], + ARRAY_SIZE(ak4535_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_input_mixer_controls[0], + ARRAY_SIZE(ak4535_input_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4535_input_mux_control), + SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4535_mono2_control), + SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0), /* speaker powersave bit */ + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ak4535_line_control), + SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpl_control), + SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpr_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPP"), + SND_SOC_DAPM_OUTPUT("SPN"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0), + SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const char *audio_map[][3] = { + /*stereo mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* mono1 mixer */ + {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Input Mux", "Internal", "Mic Int Bias"}, + {"Input Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Input Mux"}, + + /* line out */ + {"LOUT", NULL, "Line Out Enable"}, + {"ROUT", NULL, "Line Out Enable"}, + {"Line Out Enable", "Switch", "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* mono1 out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* left HP */ + {"HPL", NULL, "Left HP Enable"}, + {"Left HP Enable", "Switch", "HP L Amp"}, + {"HP L Amp", NULL, "Stereo Mixer"}, + + /* right HP */ + {"HPR", NULL, "Right HP Enable"}, + {"Right HP Enable", "Switch", "HP R Amp"}, + {"HP R Amp", NULL, "Stereo Mixer"}, + + /* speaker */ + {"SPP", NULL, "Speaker Enable"}, + {"SPN", NULL, "Speaker Enable"}, + {"Speaker Enable", "Switch", "Spk Amp"}, + {"Spk Amp", NULL, "MIN"}, + + /* mono 2 */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Stereo Mixer"}, + + /* Aux In */ + {"Aux In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "Aux In"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int ak4535_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) { + snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]); + } + + /* set up audio path audio_map interconnects */ + for(i = 0; audio_map[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + } + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4535_priv *ak4535 = codec->private_data; + + ak4535->sysclk = freq; + return 0; +} + +static int ak4535_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ak4535_priv *ak4535 = codec->private_data; + u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + int rate = params_rate(params), fs = 256; + + if (rate) + fs = ak4535->sysclk / rate; + + /* set fs */ + switch (fs) { + case 1024: + mode2 |= (0x2 << 5); + break; + case 512: + mode2 |= (0x1 << 5); + break; + case 256: + break; + } + + /* set rate */ + ak4535_write(codec, AK4535_MODE2, mode2); + return 0; +} + +static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x0002; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x0001; + break; + default: + return -EINVAL; + } + + /* use 32 fs for BCLK to save power */ + mode1 |= 0x4; + + ak4535_write(codec, AK4535_MODE1, mode1); + return 0; +} + +static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + if (!mute) + ak4535_write(codec, AK4535_DAC, mute_reg); + else + ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + return 0; +} + +static int ak4535_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 i; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + ak4535_mute(codec->dai, 0); + /* vref/mid, clk and osc on, dac unmute, active */ + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + ak4535_mute(codec->dai, 1); + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, dac mute, inactive */ + i = ak4535_read_reg_cache (codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i|0x80); + i = ak4535_read_reg_cache (codec, AK4535_PM2); + ak4535_write(codec, AK4535_PM2, i & (~0x80)); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* everything off, inactive */ + i = ak4535_read_reg_cache (codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i & (~0x80)); + break; + } + codec->dapm_state = event; + ak4535dbg_dump (codec); + return 0; +} + +#define AK4535_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) + +struct snd_soc_codec_dai ak4535_dai = { + .name = "AK4535", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = ak4535_hw_params, + }, + .dai_ops = { + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(ak4535_dai); + +static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int ak4535_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + ak4535_sync(codec); + ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + ak4535_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} + +/* + * initialise the AK4535 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4535_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "AK4535"; + codec->owner = THIS_MODULE; + codec->read = ak4535_read_reg_cache; + codec->write = ak4535_write; + codec->dapm_event = ak4535_dapm_event; + codec->dai = &ak4535_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ak4535_reg); + codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + ak4535_add_controls(codec); + ak4535_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak4535_socdev; + +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver ak4535_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = ak4535_socdev; + struct ak4535_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL){ + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = ak4535_init(socdev); + if (ret < 0) { + printk(KERN_ERR "failed to initialise AK4535\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int ak4535_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec* codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int ak4535_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, ak4535_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver ak4535_i2c_driver = { + .driver = { + .name = "AK4535 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_AK4535, + .attach_adapter = ak4535_i2c_attach, + .detach_client = ak4535_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AK4535", + .driver = &ak4535_i2c_driver, +}; +#endif + +static int ak4535_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak4535_setup_data *setup; + struct snd_soc_codec* codec; + struct ak4535_priv *ak4535; + int ret = 0; + + printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + if (ak4535 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak4535; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak4535_socdev = socdev; +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int ak4535_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec* codec = socdev->codec; + + if (codec->control_data) + ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + i2c_del_driver(&ak4535_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4535 = { + .probe = ak4535_probe, + .remove = ak4535_remove, + .suspend = ak4535_suspend, + .resume = ak4535_resume, +}; + +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); + +MODULE_DESCRIPTION("Soc AK4535 driver"); +MODULE_AUTHOR("Richard Purdie"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h new file mode 100644 index 0000000..612164e --- /dev/null +++ b/sound/soc/codecs/ak4535.h @@ -0,0 +1,46 @@ +/* + * ak4535.h -- AK4535 Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie richard@openedhand.com + * + * Based on wm8753.h + * + * 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 _AK4535_H +#define _AK4535_H + +/* AK4535 register space */ + +#define AK4535_PM1 0x0 +#define AK4535_PM2 0x1 +#define AK4535_SIG1 0x2 +#define AK4535_SIG2 0x3 +#define AK4535_MODE1 0x4 +#define AK4535_MODE2 0x5 +#define AK4535_DAC 0x6 +#define AK4535_MIC 0x7 +#define AK4535_TIMER 0x8 +#define AK4535_ALC1 0x9 +#define AK4535_ALC2 0xa +#define AK4535_PGA 0xb +#define AK4535_LATT 0xc +#define AK4535_RATT 0xd +#define AK4535_VOL 0xe +#define AK4535_STATUS 0xf + +#define AK4535_CACHEREGNUM 0x10 + +struct ak4535_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4535; + +#endif

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Giorgio Padrin giorgio@mandarinlogiq.org Signed-off-by: Richard Purdie richard@openedhand.com Signed-off-by: Philipp Zabel philipp.zabel@gmail.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com Cc: i2c@lm-sensors.org --- include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/uda1380.c | 740 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda1380.h | 89 ++++++ 5 files changed, 836 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/uda1380.c create mode 100644 sound/soc/codecs/uda1380.h

diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index f6a1ddf..85472a1 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -123,6 +123,7 @@ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_AK4535 95 /* AK4525 audio codec */ #define I2C_DRIVERID_TLV320 97 /* TLV 320 audio codec */ +#define I2C_DRIVERID_UDA1380 98 /* Philips UDA1380 audio codec */

#define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2608499..4c05f60 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,10 @@ config SND_SOC_TLV320 tristate depends on SND_SOC

+config SND_SOC_UDA1380 + tristate + depends on SND_SOC + config SND_SOC_WM8731 tristate depends on SND_SOC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3cce68b..bce9fe0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,7 @@ snd-soc-ac97-objs := ac97.o snd-soc-ak4535-objs := ak4535.o snd-soc-tlv320-objs := tlv320.o +snd-soc-uda1380-objs := uda1380.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -11,6 +12,7 @@ snd-soc-cs4270-objs := cs4270.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_TLV320) += snd-soc-tlv320.o +obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c new file mode 100644 index 0000000..0fa9832 --- /dev/null +++ b/sound/soc/codecs/uda1380.c @@ -0,0 +1,740 @@ +/* + * uda1380.c - Philips UDA1380 ALSA SoC audio driver + * + * 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. + * + * Modified by Richard Purdie richard@openedhand.com to fit into SoC + * codec model. + * + * Copyright (c) 2005 Giorgio Padrin giorgio@mandarinlogiq.org + * Copyright 2005 Openedhand Ltd. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/types.h> +#include <linux/string.h> +#include <linux/slab.h> +#include <linux/errno.h> +#include <linux/ioctl.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "uda1380.h" + +#define UDA1380_VERSION "0.5" +#define AUDIO_NAME "uda1380" +/* + * Debug + */ + +#define UDA1380_DEBUG 0 + +#ifdef UDA1380_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif + +/* + * uda1380 register cache + */ +static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { + 0x0502, 0x0000, 0x0000, 0x3f3f, + 0x0202, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0xff00, 0x0000, 0x4800, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x8000, 0x0002, 0x0000, +}; + +/* + * read uda1380 register cache + */ +static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == UDA1380_RESET) + return 0; + if (reg >= UDA1380_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write uda1380 register cache + */ +static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the UDA1380 register space + */ +static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + /* data is + * data[0] is register offset + * data[1] is MS byte + * data[2] is LS byte + */ + data[0] = reg; + data[1] = (value & 0xff00) >> 8; + data[2] = value & 0x00ff; + + uda1380_write_reg_cache (codec, reg, value); + + /* the interpolator & decimator regs must only be written when the + * codec DAI is active. + */ + if (!codec->active && (reg >= UDA1380_MVOL)) + return 0; + dbg("uda hw write %x val %x\n", reg, value); + if (codec->hw_write(codec->control_data, data, 3) == 3) { + unsigned int val; + i2c_master_send(codec->control_data, data, 1); + i2c_master_recv(codec->control_data, data, 2); + val = (data[0]<<8) | data[1]; + if (val != value) { + dbg("READ BACK VAL %x\n", (data[0]<<8) | data[1]); + return -EIO; + } + return 0; + } else + return -EIO; +} + +#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) + +/* declarations of ALSA reg_elem_REAL controls */ +static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz", + "96kHz"}; +static const char *uda1380_input_sel[] = {"Line", "Mic"}; +static const char *uda1380_output_sel[] = {"Direct", "Mixer"}; +static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2", + "Maximum"}; + +static const struct soc_enum uda1380_enum[] = { + SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel), + SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode), + SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel), /* R02_EN_AVC */ +}; + +static const struct snd_kcontrol_new uda1380_snd_controls[] = { + SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 255, 1), + SOC_DOUBLE("Mixer Volume", UDA1380_MIXVOL, 0, 8, 255, 1), + SOC_ENUM("Sound Processing Filter Mode", uda1380_enum[2]), + SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0), + SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0), + SOC_ENUM("Playback De-emphasis", uda1380_enum[0]), + SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0), + SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0), + SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0), + SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 1), + SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0), + SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), + SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), + SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), + SOC_SINGLE("Silence", UDA1380_MIXER, 7, 1, 0), + SOC_SINGLE("Silence Detection", UDA1380_MIXER, 6, 1, 0), +}; + +/* add non dapm controls */ +static int uda1380_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Input mux */ +static const struct snd_kcontrol_new uda1380_input_mux_control = + SOC_DAPM_ENUM("Input Select", uda1380_enum[1]); + +/* Output mux */ +static const struct snd_kcontrol_new uda1380_output_mux_control = + SOC_DAPM_ENUM("Output Select", uda1380_enum[3]); + +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &uda1380_input_mux_control), + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, + &uda1380_output_mux_control), + SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), + SND_SOC_DAPM_INPUT("VINM"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("VOUTLHP"), + SND_SOC_DAPM_OUTPUT("VOUTRHP"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), + SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), +}; + +static const char *audio_map[][3] = { + + /* output mux */ + {"HeadPhone Driver", NULL, "Output Mux"}, + {"VOUTR", NULL, "Output Mux"}, + {"VOUTL", NULL, "Output Mux"}, + + {"Analog Mixer", NULL, "VINR"}, + {"Analog Mixer", NULL, "VINL"}, + {"Analog Mixer", NULL, "DAC"}, + + {"Output Mux", "Direct", "DAC"}, + {"Output Mux", "Mixer", "Analog Mixer"}, + + /* headphone driver */ + {"VOUTLHP", NULL, "HeadPhone Driver"}, + {"VOUTRHP", NULL, "HeadPhone Driver"}, + + /* input mux */ + {"Left ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Line", "Left PGA"}, + + /* right input */ + {"Right ADC", NULL, "Right PGA"}, + + /* inputs */ + {"Mic LNA", NULL, "VINM"}, + {"Left PGA", NULL, "VINL"}, + {"Right PGA", NULL, "VINR"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int uda1380_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]); + + /* set up audio path interconnects */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + /* set up DAI based upon fmt */ + + iface = uda1380_read_reg_cache (codec, UDA1380_IFACE); + iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); + + /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB | R01_SFORO_I2S; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +/* + * Flush reg cache + * We can only write the interpolator and decimator registers + * when the DAI is being clocked by the CPU DAI. It's up to the + * machine and cpu DAI driver to do this before we are called. + */ +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg, reg_start, reg_end, clk; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_start = UDA1380_MVOL; + reg_end = UDA1380_MIXER; + } else { + reg_start = UDA1380_DEC; + reg_end = UDA1380_AGC; + } + + /* FIXME disable DAC_CLK */ + clk = uda1380_read_reg_cache (codec, 00); + uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); + + for (reg = reg_start; reg <= reg_end; reg++ ) { + dbg("reg %x val %x\n",reg, uda1380_read_reg_cache (codec, reg)); + uda1380_write(codec, reg, uda1380_read_reg_cache (codec, reg)); + } + + /* FIXME enable DAC_CLK */ + uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + + return 0; +} + +static int uda1380_pcm_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* set WSPLL power and divider if running from this clock */ + if (clk & R00_DAC_CLK) { + int rate = params_rate(params); + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + clk &= ~0x3; /* clear SEL_LOOP_DIV */ + switch (rate) { + case 6250 ... 12500: + clk |= 0x0; + break; + case 12501 ... 25000: + clk |= 0x1; + break; + case 25001 ... 50000: + clk |= 0x2; + break; + case 50001 ... 100000: + clk |= 0x3; + break; + } + uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk |= R00_EN_DAC | R00_EN_INT; + else + clk |= R00_EN_ADC | R00_EN_DEC; + + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* shut down WSPLL power if running from this clock */ + if (clk & R00_DAC_CLK) { + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk &= ~(R00_EN_DAC | R00_EN_INT); + else + clk &= ~(R00_EN_ADC | R00_EN_DEC); + + uda1380_write(codec, UDA1380_CLK, clk); +} + +static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff; + + /* FIXME: mute(codec,0) is called when the magician clock is already + * set to WSPLL, but for some unknown reason writing to interpolator + * registers works only when clocked by SYSCLK */ + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); + if (mute) + uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000); + else + uda1380_write(codec, UDA1380_DEEMP, mute_reg); + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static int uda1380_dapm_event(struct snd_soc_codec *codec, int event) +{ + int pm = uda1380_read_reg_cache(codec, UDA1380_PM); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + /* enable internal bias */ + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except internal bias */ + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* everything off, inactive */ + uda1380_write(codec, UDA1380_PM, 0x0); + break; + } + codec->dapm_state = event; + return 0; +} + +#define UDA1380_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) + +struct snd_soc_codec_dai uda1380_dai[] = { +{ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{/* playback only - dual interface */ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* capture only - dual interface*/ + .name = "UDA1380", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .set_fmt = uda1380_set_dai_fmt, + }, +}, +}; +EXPORT_SYMBOL_GPL(uda1380_dai); + +static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int uda1380_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + uda1380_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} + +/* + * initialise the UDA1380 driver + * register the mixer and dsp interfaces with the kernel + */ +static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->dapm_event = uda1380_dapm_event; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache_size = sizeof(uda1380_reg); + codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + uda1380_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if(ret < 0) { + printk(KERN_ERR "uda1380: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + /* set clock input */ + switch (dac_clk) { + case UDA1380_DAC_CLK_SYSCLK: + uda1380_write(codec, UDA1380_CLK, 0); + break; + case UDA1380_DAC_CLK_WSPLL: + uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK); + break; + } + + /* uda1380 init */ + uda1380_add_controls(codec); + uda1380_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "uda1380: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *uda1380_socdev; + +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver uda1380_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = uda1380_socdev; + struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL){ + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = uda1380_init(socdev, setup->dac_clk); + if (ret < 0) { + printk(KERN_ERR "failed to initialise UDA1380\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int uda1380_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec* codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int uda1380_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, uda1380_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_UDA1380, + .attach_adapter = uda1380_i2c_attach, + .detach_client = uda1380_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "UDA1380", + .driver = &uda1380_i2c_driver, +}; +#endif + +static int uda1380_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct uda1380_setup_data *setup; + struct snd_soc_codec* codec; + int ret = 0; + + printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + uda1380_socdev = socdev; +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec* codec = socdev->codec; + + if (codec->control_data) + uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, +}; + +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); + +MODULE_AUTHOR("Giorgio Padrin"); +MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h new file mode 100644 index 0000000..f9d885c --- /dev/null +++ b/sound/soc/codecs/uda1380.h @@ -0,0 +1,89 @@ +/* + * Audio support for Philips UDA1380 + * + * 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. + * + * Copyright (c) 2005 Giorgio Padrin giorgio@mandarinlogiq.org + */ + +#ifndef _UDA1380_H +#define _UDA1380_H + +#define UDA1380_CLK 0x00 +#define UDA1380_IFACE 0x01 +#define UDA1380_PM 0x02 +#define UDA1380_AMIX 0x03 +#define UDA1380_HP 0x04 +#define UDA1380_MVOL 0x10 +#define UDA1380_MIXVOL 0x11 +#define UDA1380_MODE 0x12 +#define UDA1380_DEEMP 0x13 +#define UDA1380_MIXER 0x14 +#define UDA1380_INTSTAT 0x18 +#define UDA1380_DEC 0x20 +#define UDA1380_PGA 0x21 +#define UDA1380_ADC 0x22 +#define UDA1380_AGC 0x23 +#define UDA1380_DECSTAT 0x28 +#define UDA1380_RESET 0x7f + +#define UDA1380_CACHEREGNUM 0x24 + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +struct uda1380_setup_data { + unsigned short i2c_address; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ + +extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_codec_device soc_codec_dev_uda1380; + +#endif /* _UDA1380_H */

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Nicola Perrino nicola.perrino@atlab.it Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com --- sound/soc/pxa/Kconfig | 10 ++ sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/amesom_tlv320.c | 211 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 223 insertions(+), 1 deletions(-) create mode 100644 sound/soc/pxa/amesom_tlv320.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ca825e6..bcb3aa0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -57,3 +57,13 @@ config SND_PXA2XX_SOC_TOSA help Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). + +config SND_PXA2XX_SOC_AMESOM_TLV320 + tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k" + depends on SND_PXA2XX_SOC && MACH_AMESOM + select SND_PXA2XX_SOC_I2S + select SND_PXA2XX_SOC_SSP + select SND_SOC_TLV320 + help + Say Y if you want to add support for SoC audio on Amesom + with the tlv320. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c631651..931bdc7 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -14,9 +14,10 @@ snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-spitz-objs := spitz.o +snd-soc-amesom-tlv320-objs := amesom_tlv320.o

obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o - +obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o diff --git a/sound/soc/pxa/amesom_tlv320.c b/sound/soc/pxa/amesom_tlv320.c new file mode 100644 index 0000000..6aa1c2c --- /dev/null +++ b/sound/soc/pxa/amesom_tlv320.c @@ -0,0 +1,211 @@ +/* + * amesom_tlv320.c -- SoC audio for Amesom + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2006 Atlab srl. + * + * Authors: Liam Girdwood liam.girdwood@wolfsonmicro.com + * Nicola Perrino nicola.perrino@atlab.it + * + * 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 + * option) any later version. + * + * Revision history + * 5th Dec 2006 Initial version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/hardware.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/audio.h> + +#include "../codecs/tlv320.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-ssp.h" + + +/* + * SSP2 GPIO's + */ + +#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN) +#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT) +#define GPIO50_SSP2CLKS_MD (50 | GPIO_ALT_FN_3_IN) +#define GPIO14_SSP2FRMS_MD (14 | GPIO_ALT_FN_2_IN) +#define GPIO50_SSP2CLKM_MD (50 | GPIO_ALT_FN_3_OUT) +#define GPIO14_SSP2FRMM_MD (14 | GPIO_ALT_FN_2_OUT) + + +static struct snd_soc_machine amesom; + + +static int amesom_probe(struct platform_device *pdev) +{ + return 0; +} + +static int amesom_remove(struct platform_device *pdev) +{ + return 0; +} + +static int tlv320_voice_startup(struct snd_pcm_substream *substream) +{ + return 0; +} + +static void tlv320_voice_shutdown(struct snd_pcm_substream *substream) +{ + return; +} + +/* + * Tlv320 uses SSP port for playback. + */ +static int tlv320_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + //printk("tlv320_voice_hw_params enter\n"); + switch(params_rate(params)) { + case 8000: + //printk("tlv320_voice_hw_params 8000\n"); + break; + case 16000: + //printk("tlv320_voice_hw_params 16000\n"); + break; + default: + break; + } + + // CODEC MASTER, SSP SLAVE + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the SSP system clock as input (unused) */ + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_NET_PLL, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set SSP slots */ + //ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x1, slots); + ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x3, 1); + if (ret < 0) + return ret; + + return 0; +} + +static int tlv320_voice_hw_free(struct snd_pcm_substream *substream) +{ + return 0; +} + +static struct snd_soc_ops tlv320_voice_ops = { + .startup = tlv320_voice_startup, + .shutdown = tlv320_voice_shutdown, + .hw_params = tlv320_voice_hw_params, + .hw_free = tlv320_voice_hw_free, +}; + + +static struct snd_soc_dai_link amesom_dai[] = { +{ + .name = "TLV320", + .stream_name = "TLV320 Voice", + .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2], + .codec_dai = &tlv320_dai[TLV320_DAI_MODE1_VOICE], + .ops = &tlv320_voice_ops, +}, +}; + +static struct snd_soc_machine amesom = { + .name = "Amesom", + .probe = amesom_probe, + .remove = amesom_remove, + .dai_link = amesom_dai, + .num_links = ARRAY_SIZE(amesom_dai), +}; + +static struct tlv320_setup_data amesom_tlv320_setup = { +#ifdef TLV320AIC24K //codec2 + .i2c_address = 0x41, +#else // TLV320AIC14k + .i2c_address = 0x40, +#endif +}; + +static struct snd_soc_device amesom_snd_devdata = { + .machine = &amesom, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_tlv320, + .codec_data = &amesom_tlv320_setup, +}; + +static struct platform_device *amesom_snd_device; + +static int __init amesom_init(void) +{ + int ret; + + amesom_snd_device = platform_device_alloc("soc-audio", -1); + if (!amesom_snd_device) + return -ENOMEM; + + platform_set_drvdata(amesom_snd_device, &amesom_snd_devdata); + amesom_snd_devdata.dev = &amesom_snd_device->dev; + ret = platform_device_add(amesom_snd_device); + + if (ret) + platform_device_put(amesom_snd_device); + + + /* SSP port 2 slave */ + pxa_gpio_mode(GPIO11_SSP2RX_MD); + pxa_gpio_mode(GPIO13_SSP2TX_MD); + pxa_gpio_mode(GPIO50_SSP2CLKS_MD); + pxa_gpio_mode(GPIO14_SSP2FRMS_MD); + + return ret; +} + +static void __exit amesom_exit(void) +{ + platform_device_unregister(amesom_snd_device); +} + +module_init(amesom_init); +module_exit(amesom_exit); + +/* Module information */ +MODULE_AUTHOR("Nicola Perrino"); +MODULE_DESCRIPTION("ALSA SoC TLV320 Amesom"); +MODULE_LICENSE("GPL");

From: Philipp Zabel philipp.zabel@gmail.com

Signed-off-by: Philipp Zabel philipp.zabel@gmail.com --- sound/soc/codecs/uda1380.c | 255 ++++++++++++++++++++++++++++++++++++++------ 1 files changed, 222 insertions(+), 33 deletions(-)

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 0fa9832..71960f0 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -5,6 +5,10 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * + * Copyright (c) 2007 Philipp Zabel philipp.zabel@gmail.com + * Improved support for DAPM and audio routing/mixing capabilities, + * added TLV support. + * * Modified by Richard Purdie richard@openedhand.com to fit into SoC * codec model. * @@ -28,6 +32,7 @@ #include <sound/info.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/tlv.h>

#include "uda1380.h"

@@ -111,14 +116,14 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, */ if (!codec->active && (reg >= UDA1380_MVOL)) return 0; - dbg("uda hw write %x val %x\n", reg, value); + dbg("uda hw write %x val %x", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { unsigned int val; i2c_master_send(codec->control_data, data, 1); i2c_master_recv(codec->control_data, data, 2); val = (data[0]<<8) | data[1]; if (val != value) { - dbg("READ BACK VAL %x\n", (data[0]<<8) | data[1]); + dbg("READ BACK VAL %x", (data[0]<<8) | data[1]); return -EIO; } return 0; @@ -131,37 +136,202 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, /* declarations of ALSA reg_elem_REAL controls */ static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz", "96kHz"}; -static const char *uda1380_input_sel[] = {"Line", "Mic"}; -static const char *uda1380_output_sel[] = {"Direct", "Mixer"}; +static const char *uda1380_input_sel[] = { "Line", "Mic + Line R", "Line L", "Mic" }; +static const char *uda1380_output_sel[] = {"DAC", "Analog Mixer"}; static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2", "Maximum"}; +static const char *uda1380_capture_sel[] = {"ADC", "Digital Mixer"}; +static const char *uda1380_sel_ns[] = { "3rd-order", "5th-order" }; +static const char *uda1380_mix_control[] = { "off", "PCM only", "before sound processing", "after sound processing" }; +static const char *uda1380_sdet_setting[] = { "3200", "4800", "9600", "19200" }; +static const char *uda1380_os_setting[] = { "single-speed", "double-speed (no mixing)", "quad-speed (no mixing)" }; + +static const struct soc_enum uda1380_deemp_enum[] = { + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), +}; +static const struct soc_enum uda1380_input_sel_enum = + SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static const struct soc_enum uda1380_output_sel_enum = + SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ +static const struct soc_enum uda1380_spf_enum = + SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ +static const struct soc_enum uda1380_capture_sel_enum = + SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ +static const struct soc_enum uda1380_sel_ns_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ +static const struct soc_enum uda1380_mix_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ +static const struct soc_enum uda1380_sdet_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ +static const struct soc_enum uda1380_os_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ + +/** + * snd_soc_info_volsw - single mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (kcontrol->private_value >> 16) & 0xff; + int min = (kcontrol->private_value >> 24) & 0xff; + + /* 00000000 (0) ...(-0.25 dB)... 11111000 (-78 dB), 11111100 -INF */ + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +/** + * snd_soc_get_volsw_sgn - single mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; + int min = (kcontrol->private_value >> 24) & 0xff; + + ucontrol->value.integer.value[0] = + (signed char)(snd_soc_read(codec, reg) & 0xff); + ucontrol->value.integer.value[1] = + (signed char)((snd_soc_read(codec, reg) >> 8) & 0xff); + + return 0; +} + +/** + * snd_soc_put_volsw_sgn - single mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int max = (kcontrol->private_value >> 16) & 0xff; + int min = (kcontrol->private_value >> 24) & 0xff; + int err; + unsigned short val, val2, val_mask; + + val = (signed char)ucontrol->value.integer.value[0] & 0xff; + val2 = (signed char)ucontrol->value.integer.value[1] & 0xff; + val |= val2<<8; + err = snd_soc_update_bits(codec, reg, val_mask, val); + return err; +} + +DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); /* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) */ + +static const unsigned int mvol_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0,15, TLV_DB_SCALE_ITEM(-8200, 100, 1), /* from -78 dB in 1 dB steps (3 dB steps, really) */ + 16,43, TLV_DB_SCALE_ITEM(-6600, 50, 0), /* from -66 dB in 0.5 dB steps (2 dB steps, really) */ + 44,252, TLV_DB_SCALE_ITEM(-5200, 25, 0), /* from -52 dB in 0.25 dB steps */ +};

-static const struct soc_enum uda1380_enum[] = { - SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp), - SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel), - SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode), - SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel), /* R02_EN_AVC */ +static const unsigned int vc_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0,7, TLV_DB_SCALE_ITEM(-7800, 150, 1), /* from -72 dB in 1.5 dB steps (6 dB steps really) */ + 8,15, TLV_DB_SCALE_ITEM(-6600, 75, 0), /* from -66 dB in 0.75 dB steps (3 dB steps really) */ + 16,43, TLV_DB_SCALE_ITEM(-6000, 50, 0), /* from -60 dB in 0.5 dB steps (2 dB steps really) */ + 44,228, TLV_DB_SCALE_ITEM(-4600, 25, 0),/* from -46 dB in 0.25 dB steps */ };

+DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); /* from 0 dB to 6 dB in 2 dB steps, if SPF mode != flat */ +DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); /* from 0 dB to 24 dB in 2 dB steps, if SPF mode == maximum */ + /* (SPF mode == flat cuts off at 18 dB max */ + +static const unsigned int dec_tlv[] = { + TLV_DB_RANGE_HEAD(1), + -128,48, TLV_DB_SCALE_ITEM(-6350, 50, 1), + /* 0011000 (48 dB) ...(0.5 dB)... 10000001 (-63 dB) 10000000 -INF */ +}; + +DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); /* from 0 dB to 24 dB in 3 dB steps */ +DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); /* from 0 dB to 30 dB in 2 dB steps */ + + +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | (((signed char)min) << 24) } + static const struct snd_kcontrol_new uda1380_snd_controls[] = { - SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 255, 1), - SOC_DOUBLE("Mixer Volume", UDA1380_MIXVOL, 0, 8, 255, 1), - SOC_ENUM("Sound Processing Filter Mode", uda1380_enum[2]), - SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0), - SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0), - SOC_ENUM("Playback De-emphasis", uda1380_enum[0]), - SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0), - SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0), - SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0), - SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 1), - SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0), - SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), - SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), + SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ + SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */ + SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */ + SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */ + SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */ + SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */ + SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */ +/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */ + SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */ + SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */ + SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */ + SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */ + SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ + SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ + SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ + SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ + SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ + SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ + SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ + SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */ +/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */ + SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */ + SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */ + SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */ + SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */ + SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */ + SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */ + SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */ + /* -5.5, -8, -11.5, -14 dBFS */ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), - SOC_SINGLE("Silence", UDA1380_MIXER, 7, 1, 0), - SOC_SINGLE("Silence Detection", UDA1380_MIXER, 6, 1, 0), };

+/** + * uda1380_snd_soc_cnew + * + * temporary copy of snd_soc_cnew that doesn't overwrite .access + */ +struct snd_kcontrol *uda1380_snd_soc_cnew(const struct snd_kcontrol_new *_template, + void *data, char *long_name) +{ + struct snd_kcontrol_new template; + + memcpy(&template, _template, sizeof(template)); + if (long_name) + template.name = long_name; + template.index = 0; + + return snd_ctl_new1(&template, data); +} + /* add non dapm controls */ static int uda1380_add_controls(struct snd_soc_codec *codec) { @@ -169,7 +339,7 @@ static int uda1380_add_controls(struct snd_soc_codec *codec)

for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); + uda1380_snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); if (err < 0) return err; } @@ -179,17 +349,24 @@ static int uda1380_add_controls(struct snd_soc_codec *codec)

/* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = - SOC_DAPM_ENUM("Input Select", uda1380_enum[1]); + SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);

/* Output mux */ static const struct snd_kcontrol_new uda1380_output_mux_control = - SOC_DAPM_ENUM("Output Select", uda1380_enum[3]); + SOC_DAPM_ENUM("Route", uda1380_output_sel_enum); + +/* Capture mux */ +static const struct snd_kcontrol_new uda1380_capture_mux_control = + SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum); +

static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &uda1380_input_mux_control), SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, &uda1380_output_mux_control), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &uda1380_capture_mux_control), SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), @@ -218,8 +395,10 @@ static const char *audio_map[][3] = { {"Analog Mixer", NULL, "VINL"}, {"Analog Mixer", NULL, "DAC"},

- {"Output Mux", "Direct", "DAC"}, - {"Output Mux", "Mixer", "Analog Mixer"}, + {"Output Mux", "DAC", "DAC"}, + {"Output Mux", "Analog Mixer", "Analog Mixer"}, + + /* {"DAC", "Digital Mixer", "I2S" } */

/* headphone driver */ {"VOUTLHP", NULL, "HeadPhone Driver"}, @@ -228,10 +407,13 @@ static const char *audio_map[][3] = { /* input mux */ {"Left ADC", NULL, "Input Mux"}, {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Mic LNA"}, + {"Input Mux", "Line L", "Left PGA"}, {"Input Mux", "Line", "Left PGA"},

/* right input */ - {"Right ADC", NULL, "Right PGA"}, + {"Right ADC", "Mic + Line R", "Right PGA"}, + {"Right ADC", "Line", "Right PGA"},

/* inputs */ {"Mic LNA", NULL, "VINM"}, @@ -307,6 +489,13 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) } else { reg_start = UDA1380_DEC; reg_end = UDA1380_AGC; + /* FIXME - wince values */ +// uda1380_write(codec, UDA1380_PM, uda1380_read_reg_cache (codec, UDA1380_PM) | R02_PON_LNA | R02_PON_DAC); + uda1380_write(codec, UDA1380_PM, 0x851f); +// uda1380_write_reg_cache(codec, UDA1380_DEC, 0x0000); +// uda1380_write_reg_cache(codec, UDA1380_PGA, uda1380_read_reg_cache (codec, UDA1380_PM) & ~R21_MT_ADC); /* unmute */ +// uda1380_write_reg_cache(codec, UDA1380_ADC, 0x0001/*0x090d*/); + // ADC_POL_INV=0 VGA_CTRL=9(1001:18dB) gain SEL_LNA=1 SEL_MIC=1 EN_DCFIL=1 }

/* FIXME disable DAC_CLK */ @@ -314,7 +503,7 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);

for (reg = reg_start; reg <= reg_end; reg++ ) { - dbg("reg %x val %x\n",reg, uda1380_read_reg_cache (codec, reg)); + dbg("flush reg %x val %x:",reg, uda1380_read_reg_cache (codec, reg)); uda1380_write(codec, reg, uda1380_read_reg_cache (codec, reg)); }

@@ -387,7 +576,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; - u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff; + u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;

/* FIXME: mute(codec,0) is called when the magician clock is already * set to WSPLL, but for some unknown reason writing to interpolator @@ -395,7 +584,7 @@ static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); if (mute) - uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000); + uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); else uda1380_write(codec, UDA1380_DEEMP, mute_reg); uda1380_write(codec, UDA1380_CLK, clk);

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com Cc: i2c@lm-sensors.org --- include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8711.c | 707 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8711.h | 42 +++ 5 files changed, 756 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h

diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index fbbedbd..7765c6c 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -125,6 +125,7 @@ #define I2C_DRIVERID_TLV320 97 /* TLV 320 audio codec */ #define I2C_DRIVERID_UDA1380 98 /* Philips UDA1380 audio codec */ #define I2C_DRIVERID_WM8510 99 /* Wolfson WM8510 audio codec */ +#define I2C_DRIVERID_WM8711 100 /* Wolfson WM8711 audio codec */

#define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4e8add6..9a0ec07 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,10 @@ config SND_SOC_WM8510 tristate depends on SND_SOC

+config SND_SOC_WM8711 + tristate + depends on SND_SOC + config SND_SOC_WM8731 tristate depends on SND_SOC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a60778f..b6f8578 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-tlv320-objs := tlv320.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o +snd-soc-wm8711-objs := wm8711.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_TLV320) += snd-soc-tlv320.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o +obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c new file mode 100644 index 0000000..8631297 --- /dev/null +++ b/sound/soc/codecs/wm8711.c @@ -0,0 +1,707 @@ +/* + * wm8711.c -- WM8711 ALSA SoC Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur linux@wolfsonmicro.com + * + * Based on wm8731.c by Richard Purdie + * + * 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 <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8711.h" + +#define AUDIO_NAME "wm8711" +#define WM8711_VERSION "0.3" + +/* + * Debug + */ + +#define WM8711_DEBUG 0 + +#ifdef WM8711_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8711; + +/* codec private data */ +struct wm8711_priv { + unsigned int sysclk; +}; + +/* + * wm8711 register cache + * We can't read the WM8711 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { + 0x0079, 0x0079, 0x000a, 0x0008, + 0x009f, 0x000a, 0x0000, 0x0000 +}; + +/* + * read wm8711 register cache + */ +static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8711_RESET) + return 0; + if (reg >= WM8711_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8711 register cache + */ +static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8711_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8711 register space + */ +static int wm8711_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8753 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8711_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8711_reset(c) wm8711_write(c, WM8711_RESET, 0) + +static const struct snd_kcontrol_new wm8711_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, + 7, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8711_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8711_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const snd_kcontrol_new_t wm8711_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, + &wm8711_output_mixer_controls[0], + ARRAY_SIZE(wm8711_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const char *intercon[][3] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int wm8711_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm8711_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm8711_dapm_widgets[i]); + + /* set up audio path interconnects */ + for (i = 0; intercon[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, intercon[i][0], + intercon[i][1], intercon[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int wm8711_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct wm8711_priv *wm8711 = codec->private_data; + u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc; + int i = get_coeff(wm8711->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + wm8711_write(codec, WM8711_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + wm8711_write(codec, WM8711_IFACE, iface); + return 0; +} + +static int wm8711_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + wm8711_write(codec, WM8711_ACTIVE, 0x0001); + return 0; +} + +static void wm8711_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + wm8711_write(codec, WM8711_ACTIVE, 0x0); + } +} + +static int wm8711_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7; + + if (mute) + wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8); + else + wm8711_write(codec, WM8711_APDIGI, mute_reg); + + return 0; +} + +static int wm8711_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8711->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8711_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + wm8711_write(codec, WM8711_IFACE, iface); + return 0; +} + + +static int wm8711_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* vref/mid, osc on, dac unmute */ + wm8711_write(codec, WM8711_PWR, reg); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, */ + wm8711_write(codec, WM8711_PWR, reg | 0x0040); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* everything off, dac mute, inactive */ + wm8711_write(codec, WM8711_ACTIVE, 0x0); + wm8711_write(codec, WM8711_PWR, 0xffff); + break; + } + codec->dapm_state = event; + return 0; +} + +#define WM8711_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + +#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_codec_dai wm8711_dai = { + .name = "WM8711", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8711_RATES, + .formats = WM8711_FORMATS,}, + .ops = { + .prepare = wm8711_pcm_prepare, + .hw_params = wm8711_hw_params, + .shutdown = wm8711_shutdown, + }, + .dai_ops = { + .digital_mute = wm8711_mute, + .set_sysclk = wm8711_set_dai_sysclk, + .set_fmt = wm8711_set_dai_fmt, + }, +}; +EXPORT_SYMBOL_GPL(wm8711_dai); + +static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8711_write(codec, WM8711_ACTIVE, 0x0); + wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int wm8711_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8711_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} + +/* + * initialise the WM8711 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8711_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "WM8711"; + codec->owner = THIS_MODULE; + codec->read = wm8711_read_reg_cache; + codec->write = wm8711_write; + codec->dapm_event = wm8711_dapm_event; + codec->dai = &wm8711_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(wm8711_reg); + codec->reg_cache = kmemdup(wm8711_reg, sizeof(wm8711_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8711_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8711: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* set the update bits */ + reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); + wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); + wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); + + wm8711_add_controls(codec); + wm8711_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8711: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8711_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8711 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8711_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8711_socdev; + struct wm8711_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + + i2c_set_clientdata(i2c, codec); + + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8711_init(socdev); + if (ret < 0) { + err("failed to initialise WM8711\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8711_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8711_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8711_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8711_i2c_driver = { + .driver = { + .name = "WM8711 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8711, + .attach_adapter = wm8711_i2c_attach, + .detach_client = wm8711_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8711", + .driver = &wm8711_i2c_driver, +}; +#endif + +static int wm8711_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8711_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8711_priv *wm8711; + int ret = 0; + + info("WM8711 Audio Codec %s", WM8711_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8711; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8711_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8711_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8711_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8711_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8711 = { + .probe = wm8711_probe, + .remove = wm8711_remove, + .suspend = wm8711_suspend, + .resume = wm8711_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); + +MODULE_DESCRIPTION("ASoC WM8711 driver"); +MODULE_AUTHOR("Mike Arthur"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h new file mode 100644 index 0000000..05ef13f --- /dev/null +++ b/sound/soc/codecs/wm8711.h @@ -0,0 +1,42 @@ +/* + * wm8711.h -- WM8711 Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur linux@wolfsonmicro.com + * + * Based on wm8731.h + * + * 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 _WM8711_H +#define _WM8711_H + +/* WM8711 register space */ + +#define WM8711_LOUT1V 0x02 +#define WM8711_ROUT1V 0x03 +#define WM8711_APANA 0x04 +#define WM8711_APDIGI 0x05 +#define WM8711_PWR 0x06 +#define WM8711_IFACE 0x07 +#define WM8711_SRATE 0x08 +#define WM8711_ACTIVE 0x09 +#define WM8711_RESET 0x0f + +#define WM8711_CACHEREGNUM 8 + +#define WM8711_SYSCLK 0 +#define WM8711_DAI 0 + +struct wm8711_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8711_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8711; + +#endif

At Tue, 20 Nov 2007 09:25:43 +0000, Mark Brown wrote:

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Philipp Zabel philipp.zabel@gmail.com Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com

Hmm... Some patches seem to have inconsistent From: and the author attribute. For exmaple, this module has

MODULE_AUTHOR("Philipp Zabel");

while Liam is in From header here (and it's a broken address :)

Otherwise it looks OK except for a few coding style issues. To be sure, try checkpatch.pl.

Takashi

---

sound/soc/pxa/Kconfig | 11 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 539 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 552 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/magician.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index bcb3aa0..3682f38 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -58,6 +58,17 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa).

+config SND_PXA2XX_SOC_MAGICIAN

• tristate "SoC Audio support for HTC Magician"
• depends on SND_PXA2XX_SOC
• select SND_PXA2XX_SOC_I2S
• select SND_PXA2XX_SOC_SSP
• select SND_SOC_UDA1380
• help

•  Say Y if you want to add support for SoC audio on the
  

•  HTC Magician.
  

config SND_PXA2XX_SOC_AMESOM_TLV320 tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k" depends on SND_PXA2XX_SOC && MACH_AMESOM diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 931bdc7..1faa751 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -15,9 +15,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-spitz-objs := spitz.o snd-soc-amesom-tlv320-objs := amesom_tlv320.o +snd-soc-magician-objs := magician.o

obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o \ No newline at end of file diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 0000000..7eb671c --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,539 @@ +/*

• • SoC audio for HTC Magician
• • Copyright (c) 2006 Philipp Zabel philipp.zabel@gmail.com
• • based on spitz.c,
• • Authors: Liam Girdwood liam.girdwood@wolfsonmicro.com

• •      Richard Purdie <richard@openedhand.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; either version 2 of the License, or (at your
• • option) any later version.
• */

+#include <linux/module.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h>

+#include <asm/hardware/scoop.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/hardware.h> +#include <asm/arch/magician.h> +#include <asm/arch/magician_cpld.h> +#include <asm/mach-types.h> +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-ssp.h"

+#define MAGICIAN_HP_ON 0 +#define MAGICIAN_HP_OFF 1

+#define MAGICIAN_SPK_ON 0 +#define MAGICIAN_SPK_OFF 1

+#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1

+/*

• • SSP GPIO's
• */

+#define GPIO23_SSPSCLK_MD (23 | GPIO_ALT_FN_2_OUT) +#define GPIO24_SSPSFRM_MD (24 | GPIO_ALT_FN_2_OUT) +#define GPIO25_SSPTXD_MD (25 | GPIO_ALT_FN_2_OUT)

+static int magician_hp_func = MAGICIAN_HP_OFF; +static int magician_spk_func = MAGICIAN_SPK_ON; +static int magician_in_sel = MAGICIAN_MIC;

+extern struct platform_device magician_cpld;

+static void magician_ext_control(struct snd_soc_codec *codec) +{

• snd_soc_dapm_set_endpoint(codec, "Speaker",

• 	(magician_spk_func == MAGICIAN_SPK_ON));
  

• snd_soc_dapm_set_endpoint(codec, "Headphone Jack",

• 	(magician_hp_func == MAGICIAN_HP_ON));
  

• switch (magician_in_sel) {
• case MAGICIAN_MIC:

• snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
  

• snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
  

• break;
  

• case MAGICIAN_MIC_EXT:

• snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
  

• snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
  

• break;
  

• }
• snd_soc_dapm_sync_endpoints(codec);

+}

+static int magician_startup(struct snd_pcm_substream *substream) +{

• struct snd_soc_pcm_runtime *rtd = substream->private_data;
• struct snd_soc_codec *codec = rtd->socdev->codec;
• /* check the jack status at stream startup */
• magician_ext_control(codec);
• return 0;

+}

+/*

• • Magician uses SSP port for playback.
• */

+static int magician_playback_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_codec_dai *codec_dai = rtd->dai->codec_dai;
• struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
• unsigned int acps, acds, div4;
• int ret = 0;
• /*

• * Rate = SSPSCLK / (word size(16))
  

• * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
  

• */
  

• switch (params_rate(params)) {
• case 8000:

• acps = 32842000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_32;	/* wrong - 32 bits/sample */
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 11025:

• acps = 5622000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_8;	/* 16 bits/sample, 1 slot */
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 22050:

• acps = 5622000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 44100:

• acps = 11345000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• case 48000:

• acps = 12235000;
  

• acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
  

• div4 = PXA2XX_SSP_CLK_SCDB_4;
  

• break;
  

• }
• /* set codec DAI configuration */
• ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |

• 	SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  

• if (ret < 0)

• return ret;
  

• /* set cpu DAI configuration */
• ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |

• 	SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  

• if (ret < 0)

• return ret;
  

• /* set audio clock as clock source */
• ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,

• 	SND_SOC_CLOCK_OUT);
  

• if (ret < 0)

• return ret;
  

• /* set the SSP audio system clock ACDS divider */
• ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,

• 	PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
  

• if (ret < 0)

• return ret;
  

• /* set the SSP audio system clock SCDB divider4 */
• ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,

• 	PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
  

• if (ret < 0)

• return ret;
  

• /* set SSP audio pll clock */
• ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
• if (ret < 0)

• return ret;
  

• return 0;

+}

+/*

• • We have to enable the SSP port early so the UDA1380 can flush
• • it's register cache. The UDA1380 can only write it's interpolator and
• • decimator registers when the link is running.
• */

+static int magician_playback_prepare(struct snd_pcm_substream *substream) +{

• /* enable SSP clock - is this needed ? */
• SSCR0_P(1) |= SSCR0_SSE;
• /* FIXME: ENABLE I2S */
• SACR0 |= SACR0_BCKD;
• SACR0 |= SACR0_ENB;
• pxa_set_cken(CKEN8_I2S, 1);
• return 0;

+}

+static int magician_playback_hw_free(struct snd_pcm_substream *substream) +{

• /* FIXME: DISABLE I2S */
• SACR0 &= ~SACR0_ENB;
• SACR0 &= ~SACR0_BCKD;
• pxa_set_cken(CKEN8_I2S, 0);
• return 0;

+}

+/*

• • Magician uses I2S for capture.
• */

+static int magician_capture_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_codec_dai *codec_dai = rtd->dai->codec_dai;
• struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
• int ret = 0;
• /* set codec DAI configuration */
• ret = codec_dai->dai_ops.set_fmt(codec_dai,

• 	SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  

• if (ret < 0)

• return ret;
  

• /* set cpu DAI configuration */
• ret = cpu_dai->dai_ops.set_fmt(cpu_dai,

• 	SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  

• if (ret < 0)

• return ret;
  

• /* set the I2S system clock as output */
• ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,

• 	SND_SOC_CLOCK_OUT);
  

• if (ret < 0)

• return ret;
  

• return 0;

+}

+/*

• • We have to enable the I2S port early so the UDA1380 can flush
• • it's register cache. The UDA1380 can only write it's interpolator and
• • decimator registers when the link is running.
• */

+static int magician_capture_prepare(struct snd_pcm_substream *substream) +{

• SACR0 |= SACR0_ENB;
• return 0;

+}

+static struct snd_soc_ops magician_capture_ops = {

• .startup = magician_startup,
• .hw_params = magician_capture_hw_params,
• .prepare = magician_capture_prepare,

+};

+static struct snd_soc_ops magician_playback_ops = {

• .startup = magician_startup,
• .hw_params = magician_playback_hw_params,
• .prepare = magician_playback_prepare,
• .hw_free = magician_playback_hw_free,

+};

+static int magician_get_jack(struct snd_kcontrol * kcontrol,

• 	     struct snd_ctl_elem_value * ucontrol)
  

+{

• ucontrol->value.integer.value[0] = magician_hp_func;
• return 0;

+}

+static int magician_set_hp(struct snd_kcontrol * kcontrol,

• 	     struct snd_ctl_elem_value * ucontrol)
  

+{

• struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
• if (magician_hp_func == ucontrol->value.integer.value[0])

• return 0;
  

• magician_hp_func = ucontrol->value.integer.value[0];
• magician_ext_control(codec);
• return 1;

+}

+static int magician_get_spk(struct snd_kcontrol * kcontrol,

• 	    struct snd_ctl_elem_value * ucontrol)
  

+{

• ucontrol->value.integer.value[0] = magician_spk_func;
• return 0;

+}

+static int magician_set_spk(struct snd_kcontrol * kcontrol,

• 	    struct snd_ctl_elem_value * ucontrol)
  

+{

• struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
• if (magician_spk_func == ucontrol->value.integer.value[0])

• return 0;
  

• magician_spk_func = ucontrol->value.integer.value[0];
• magician_ext_control(codec);
• return 1;

+}

+static int magician_get_input(struct snd_kcontrol * kcontrol,

• 	      struct snd_ctl_elem_value * ucontrol)
  

+{

• ucontrol->value.integer.value[0] = magician_in_sel;
• return 0;

+}

+static int magician_set_input(struct snd_kcontrol * kcontrol,

• 	      struct snd_ctl_elem_value * ucontrol)
  

+{

• struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
• if (magician_in_sel == ucontrol->value.integer.value[0])

• return 0;
  

• magician_in_sel = ucontrol->value.integer.value[0];
• switch (magician_in_sel) {
• case MAGICIAN_MIC:

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL0);
  

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_IN_SEL1);
  

• break;
  

• case MAGICIAN_MIC_EXT:

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL0);
  

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_IN_SEL1);
  

• }
• return 1;

+}

+static int magician_spk_power(struct snd_soc_dapm_widget *w, int event) +{

• if (SND_SOC_DAPM_EVENT_ON(event))

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_SPK_POWER);
  

• else

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_SPK_POWER);
  

• return 0;

+}

+static int magician_hp_power(struct snd_soc_dapm_widget *w, int event) +{

• if (SND_SOC_DAPM_EVENT_ON(event))

• magician_egpio_enable(&magician_cpld,
  

• 		      EGPIO_NR_MAGICIAN_EP_POWER);
  

• else

• magician_egpio_disable(&magician_cpld,
  

• 		       EGPIO_NR_MAGICIAN_EP_POWER);
  

• return 0;

+}

+static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event) +{ +// if (SND_SOC_DAPM_EVENT_ON(event)) +// magician_egpio_enable(&magician_cpld, +// EGPIO_NR_MAGICIAN_MIC_POWER); +// else +// magician_egpio_disable(&magician_cpld, +// EGPIO_NR_MAGICIAN_MIC_POWER);

• return 0;

+}

+/* magician machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {

• SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
• SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
• SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
• SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),

+};

+/* magician machine audio_map */ +static const char *audio_map[][3] = {

• /* Headphone connected to VOUTL, VOUTR */
• {"Headphone Jack", NULL, "VOUTL"},
• {"Headphone Jack", NULL, "VOUTR"},
• /* Speaker connected to VOUTL, VOUTR */
• {"Speaker", NULL, "VOUTL"},
• {"Speaker", NULL, "VOUTR"},
• /* Mics are connected to VINM */
• {"VINM", NULL, "Headset Mic"},
• {"VINM", NULL, "Call Mic"},
• {NULL, NULL, NULL},

+};

+static const char *hp_function[] = { "On", "Off" }; +static const char *spk_function[] = { "On", "Off" }; +static const char *input_select[] = { "Call Mic", "Headset Mic" }; +static const struct soc_enum magician_enum[] = {

• SOC_ENUM_SINGLE_EXT(4, hp_function),
• SOC_ENUM_SINGLE_EXT(2, spk_function),
• SOC_ENUM_SINGLE_EXT(2, input_select),

+};

+static const struct snd_kcontrol_new uda1380_magician_controls[] = {

• SOC_ENUM_EXT("Headphone Switch", magician_enum[0], magician_get_jack,

• 	magician_set_hp),
  

• SOC_ENUM_EXT("Speaker Switch", magician_enum[1], magician_get_spk,

• 	magician_set_spk),
  

• SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input,

• 	magician_set_input),
  

+};

+/*

• • Logic for a uda1380 as connected on a HTC Magician
• */

+static int magician_uda1380_init(struct snd_soc_codec *codec) +{

• int i, err;
• /* NC codec pins */
• snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
• snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
• /* FIXME: is anything connected here? */
• snd_soc_dapm_set_endpoint(codec, "VINL", 0);
• snd_soc_dapm_set_endpoint(codec, "VINR", 0);
• /* Add magician specific controls */
• for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {

• if ((err = snd_ctl_add(codec->card,
  

• 		snd_soc_cnew(&uda1380_magician_controls[i],
  

• 		codec, NULL))) < 0)
  

• 	return err;
  

• }
• /* Add magician specific widgets */
• for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {

• snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
  

• }
• /* Set up magician specific audio path interconnects */
• for (i = 0; audio_map[i][0] != NULL; i++) {

• snd_soc_dapm_connect_input(codec, audio_map[i][0],
  

• 		audio_map[i][1], audio_map[i][2]);
  

• }
• snd_soc_dapm_sync_endpoints(codec);
• return 0;

+}

+/* magician digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link magician_dai[] = { +{

• .name = "uda1380",
• .stream_name = "UDA1380 Playback",
• .cpu_dai = &pxa_ssp_dai[0],
• .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
• .init = magician_uda1380_init,
• .ops = &magician_playback_ops,

+}, +{

• .name = "uda1380",
• .stream_name = "UDA1380 Capture",
• .cpu_dai = &pxa_i2s_dai,
• .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
• .ops = &magician_capture_ops,

+} +};

+/* magician audio machine driver */ +static struct snd_soc_machine snd_soc_machine_magician = {

• .name = "Magician",
• .dai_link = magician_dai,
• .num_links = ARRAY_SIZE(magician_dai),

+};

+/* magician audio private data */ +static struct uda1380_setup_data magician_uda1380_setup = {

• .i2c_address = 0x18,
• .dac_clk = UDA1380_DAC_CLK_WSPLL,

+};

+/* magician audio subsystem */ +static struct snd_soc_device magician_snd_devdata = {

• .machine = &snd_soc_machine_magician,
• .platform = &pxa2xx_soc_platform,
• .codec_dev = &soc_codec_dev_uda1380,
• .codec_data = &magician_uda1380_setup,

+};

+static struct platform_device *magician_snd_device;

+static int __init magician_init(void) +{

• int ret;
• if (!machine_is_magician())

• return -ENODEV;
  

• magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
• /* we may need to have the clock running here - pH5 */
• magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
• udelay(5);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
• /* correct place? we'll need it to talk to the uda1380 */
• request_module("i2c-pxa");
• magician_snd_device = platform_device_alloc("soc-audio", -1);
• if (!magician_snd_device)

• return -ENOMEM;
  

• platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
• magician_snd_devdata.dev = &magician_snd_device->dev;
• ret = platform_device_add(magician_snd_device);
• if (ret)

• platform_device_put(magician_snd_device);
  

• pxa_gpio_mode(GPIO23_SSPSCLK_MD);
• pxa_gpio_mode(GPIO24_SSPSFRM_MD);
• pxa_gpio_mode(GPIO25_SSPTXD_MD);
• return ret;

+}

+static void __exit magician_exit(void) +{

• platform_device_unregister(magician_snd_device);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
• magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);

+}

+module_init(magician_init); +module_exit(magician_exit);

+MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician");

+MODULE_LICENSE("GPL");

1.5.3.5

---

Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

Mark,

I suggest to do the same naming changes in the machine part like I 've suggested for tlv32024K codec driver: rename all tlv320 places to tlv320aic2x and "aic2x_"

Also I suggest to name all static functions in the machine ASoC part to begin from ameson_ and not the codec one.

Regards, Vladimir

Mark Brown wrote:

From: Liam Girdwood liam@localhost.localdomain

Signed-off-by: Nicola Perrino nicola.perrino@atlab.it Signed-off-by: Liam Girdwood lg@opensource.wolfsonmicro.com

---

sound/soc/pxa/Kconfig | 10 ++ sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/amesom_tlv320.c | 211 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 223 insertions(+), 1 deletions(-) create mode 100644 sound/soc/pxa/amesom_tlv320.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ca825e6..bcb3aa0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -57,3 +57,13 @@ config SND_PXA2XX_SOC_TOSA help Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa).

+config SND_PXA2XX_SOC_AMESOM_TLV320

• tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k"
• depends on SND_PXA2XX_SOC && MACH_AMESOM
• select SND_PXA2XX_SOC_I2S
• select SND_PXA2XX_SOC_SSP
• select SND_SOC_TLV320
• help

•  Say Y if you want to add support for SoC audio on Amesom
  

•  with the tlv320.
  

diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c631651..931bdc7 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -14,9 +14,10 @@ snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-spitz-objs := spitz.o +snd-soc-amesom-tlv320-objs := amesom_tlv320.o

obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o

+obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o diff --git a/sound/soc/pxa/amesom_tlv320.c b/sound/soc/pxa/amesom_tlv320.c new file mode 100644 index 0000000..6aa1c2c --- /dev/null +++ b/sound/soc/pxa/amesom_tlv320.c @@ -0,0 +1,211 @@ +/*

• • amesom_tlv320.c -- SoC audio for Amesom
• • Copyright 2005 Wolfson Microelectronics PLC.
• • Copyright 2006 Atlab srl.
• • Authors: Liam Girdwood liam.girdwood@wolfsonmicro.com

• •      Nicola Perrino <nicola.perrino@atlab.it>
    

• • 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
• • option) any later version.
• • Revision history
• • 5th Dec 2006 Initial version.
• */

+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h>

+#include <asm/hardware.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/audio.h>

+#include "../codecs/tlv320.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-ssp.h"

+/*

• • SSP2 GPIO's
• */

+#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN) +#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT) +#define GPIO50_SSP2CLKS_MD (50 | GPIO_ALT_FN_3_IN) +#define GPIO14_SSP2FRMS_MD (14 | GPIO_ALT_FN_2_IN) +#define GPIO50_SSP2CLKM_MD (50 | GPIO_ALT_FN_3_OUT) +#define GPIO14_SSP2FRMM_MD (14 | GPIO_ALT_FN_2_OUT)

+static struct snd_soc_machine amesom;

+static int amesom_probe(struct platform_device *pdev) +{

• return 0;

+}

+static int amesom_remove(struct platform_device *pdev) +{

• return 0;

+}

+static int tlv320_voice_startup(struct snd_pcm_substream *substream) +{

• return 0;

+}

+static void tlv320_voice_shutdown(struct snd_pcm_substream *substream) +{

• return;

+}

+/*

• • Tlv320 uses SSP port for playback.
• */

+static int tlv320_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
• struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
• int ret = 0;
• //printk("tlv320_voice_hw_params enter\n");
• switch(params_rate(params)) {
• case 8000:

• //printk("tlv320_voice_hw_params 8000\n");
  

• break;
  

• case 16000:

• //printk("tlv320_voice_hw_params 16000\n");
  

• break;
  

• default:

• break;
  

• }
• // CODEC MASTER, SSP SLAVE
• /* set codec DAI configuration */
• ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |

• SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
  

• if (ret < 0)

• return ret;
  

• /* set cpu DAI configuration */
• ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |

• SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
  

• if (ret < 0)

• return ret;
  

• /* set the SSP system clock as input (unused) */
• ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_NET_PLL, 0,

• SND_SOC_CLOCK_IN);
  

• if (ret < 0)

• return ret;
  

• /* set SSP slots */
• //ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x1, slots);
• ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x3, 1);
• if (ret < 0)

• return ret;
  

• return 0;

+}

+static int tlv320_voice_hw_free(struct snd_pcm_substream *substream) +{

• return 0;

+}

+static struct snd_soc_ops tlv320_voice_ops = {

• .startup = tlv320_voice_startup,
• .shutdown = tlv320_voice_shutdown,
• .hw_params = tlv320_voice_hw_params,
• .hw_free = tlv320_voice_hw_free,

+};

+static struct snd_soc_dai_link amesom_dai[] = { +{

• .name = "TLV320",
• .stream_name = "TLV320 Voice",
• .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
• .codec_dai = &tlv320_dai[TLV320_DAI_MODE1_VOICE],
• .ops = &tlv320_voice_ops,

+}, +};

+static struct snd_soc_machine amesom = {

• .name = "Amesom",
• .probe = amesom_probe,
• .remove = amesom_remove,
• .dai_link = amesom_dai,
• .num_links = ARRAY_SIZE(amesom_dai),

+};

+static struct tlv320_setup_data amesom_tlv320_setup = { +#ifdef TLV320AIC24K //codec2

• .i2c_address = 0x41,

+#else // TLV320AIC14k

• .i2c_address = 0x40,

+#endif +};

+static struct snd_soc_device amesom_snd_devdata = {

• .machine = &amesom,
• .platform = &pxa2xx_soc_platform,
• .codec_dev = &soc_codec_dev_tlv320,
• .codec_data = &amesom_tlv320_setup,

+};

+static struct platform_device *amesom_snd_device;

+static int __init amesom_init(void) +{

• int ret;
• amesom_snd_device = platform_device_alloc("soc-audio", -1);
• if (!amesom_snd_device)

• return -ENOMEM;
  

• platform_set_drvdata(amesom_snd_device, &amesom_snd_devdata);
• amesom_snd_devdata.dev = &amesom_snd_device->dev;
• ret = platform_device_add(amesom_snd_device);
• if (ret)

• platform_device_put(amesom_snd_device);
  

• /* SSP port 2 slave */
• pxa_gpio_mode(GPIO11_SSP2RX_MD);
• pxa_gpio_mode(GPIO13_SSP2TX_MD);
• pxa_gpio_mode(GPIO50_SSP2CLKS_MD);
• pxa_gpio_mode(GPIO14_SSP2FRMS_MD);
• return ret;

+}

+static void __exit amesom_exit(void) +{

• platform_device_unregister(amesom_snd_device);

+}

+module_init(amesom_init); +module_exit(amesom_exit);

+/* Module information */ +MODULE_AUTHOR("Nicola Perrino"); +MODULE_DESCRIPTION("ALSA SoC TLV320 Amesom"); +MODULE_LICENSE("GPL");

At Tue, 20 Nov 2007 09:25:39 +0000, Mark Brown wrote:

diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index 5ced329..f6a1ddf 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -122,6 +122,7 @@ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_AK4535 95 /* AK4525 audio codec */ +#define I2C_DRIVERID_TLV320 97 /* TLV 320 audio codec */

Where is 96? :-)

--- /dev/null +++ b/sound/soc/codecs/tlv320.c

(snip)

+//#define TLV320_DEBUG 0

Avoid C++ style comments. Use C-style /**/ instead.

+#define TLV320_VOICE_BITS \

• (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
• SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)

Double-check if it's really S24_LE or S24_3LE.

+static int caps_charge = 2000; +static int setting = 1; +module_param(caps_charge, int, 0); +module_param(setting, int, 0);

No description for setting variable? Also, this should be bool instead of int.

+/* ADDR table */ +static const unsigned char tlv320_reg_addr[] = {

• 0x00, /* CONTROL REG 0 No Operation */
• 0x01, /* CONTROL REG 1 */

•    0x02,	/* CONTROL REG 2  */
  

•    0x03,	/* CONTROL REG 3A */
  

Spaces and tabs are mixed up. Use tabs please.

+static inline unsigned int tlv320_read_reg_cache(struct snd_soc_codec *codec,

• unsigned int reg)

+{

• u8 *cache = codec->reg_cache;
• if (reg > ARRAY_SIZE(tlv320_reg_addr))

Shouldn't it be "reg >= ARRAY_SIZE(tlv_320_reg_addr)" ?

+static inline void tlv320_write_reg_cache(struct snd_soc_codec *codec,

• unsigned int reg, unsigned int value)

+{

• u8 *cache = codec->reg_cache;
• if (reg > ARRAY_SIZE(tlv320_reg_addr))

• return;
  

Ditto.

+static int tlv320_write(struct snd_soc_codec *codec, unsigned int reg,

• unsigned int value)

+{

• if (tlv320_reg_addr[reg] > 0x06)

• return -1;
  

• tlv320_write_reg_cache (codec, reg, value);
• return i2c_smbus_write_byte_data(codec->control_data, tlv320_reg_addr[reg], value);

+/*

• • write block tlv320
• */

+static int tlv320_write_block (struct snd_soc_codec *codec,

• const u8 *data, unsigned int len)

+{

• int ret = -1;
• int i;
• for (i=0; i<len; i++) {

• dbg("addr = 0x%02x, data = 0x%02x", tlv320_reg_addr[i], data[i]);
  

• if ((ret = tlv320_write(codec, i, data[i])) < 0)
  

Avoid if ((xxx = yyy()) < 0) style. This isn't recommended any more. Split to two lines, xxx = yyy(); if (xxx < 0) ...

+static int tlv320_init(struct snd_soc_device *socdev) +{

...

• /* register pcms */
• ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);

SNDRV_DEFAULT_IDX1 and SNDRV_DEFAULT_STR1 are for card index and name string, not for PCM instance. Pass 0 and any suitable string here.

• queue_delayed_work(tlv320_workq,

• &codec->delayed_work, msecs_to_jiffies(caps_charge));
  

Remove this unless you really implement the workq stuff. (Ditto for workq creation.) Creation of a workq isn't no cost even if it's not used.

Also, there are some other coding style issues. Try checkpatch.pl to find out them.

Furthermore, if this driver supports only I2C right now, it'd be better to add kconfig dependency to CONFIG_I2C.

Takashi

Hi Takashi, Mark,

On Tue, 20 Nov 2007 11:53:25 +0100, Takashi Iwai wrote:

At Tue, 20 Nov 2007 09:25:39 +0000, Mark Brown wrote:

...

diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index 5ced329..f6a1ddf 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -122,6 +122,7 @@ #define I2C_DRIVERID_VP27SMPX 93 /* Panasonic VP27s tuner internal MPX */ #define I2C_DRIVERID_CS4270 94 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_AK4535 95 /* AK4525 audio codec */ +#define I2C_DRIVERID_TLV320 97 /* TLV 320 audio codec */

Where is 96? :-)

You don't seem to use these driver IDs anywhere? These IDs are optional, so if you have no need for them, it's much easier to just not define them.

Note that the i2c_driver.id field is planed for removal anyway.

At Tue, 20 Nov 2007 09:25:38 +0000, Mark Brown wrote:

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0675ef..8244107 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,10 @@ config SND_SOC_AC97_CODEC tristate depends on SND_SOC

+config SND_SOC_AK4535

• tristate
• depends on SND_SOC

Unless you add another interface support, this driver seems only for I2C. Then let's add dependency on I2C (which will avoid a possible unresolved symbol error).

--- /dev/null +++ b/sound/soc/codecs/ak4535.c

ak4535.c has some obvious coding-style issues. Try to run scripts/checkpatch.pl and fix the reported parts.

Takashi

At Tue, 20 Nov 2007 09:25:35 +0000, Mark Brown wrote:

---

include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

No author and sign-off? It's trivial so I don't care much, though.

Takashi

diff --git a/include/sound/soc.h b/include/sound/soc.h index e32fa2f..1c22931 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -22,7 +22,7 @@ #include <sound/control.h> #include <sound/ac97_codec.h>

-#define SND_SOC_VERSION "0.13.1" +#define SND_SOC_VERSION "0.13.2"

/*

• Convenience kcontrol builders

-- 1.5.3.5

---

Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

At Tue, 20 Nov 2007 09:25:30 +0000, Mark Brown wrote:

From: Liam Girdwood liam@localhost.localdomain

Looks suspicious. Also, sign-off is missing.

Takashi

---

sound/soc/soc-core.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 265bd55..f3e1b74 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -288,6 +288,14 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) {

• 	/* power down the codec to D1 if no longer active */
  

• 	if (codec->active == 0) {
  

• 		dbg("pop wq D1 %s %s\n", codec->name,
  

• 			codec_dai->playback.stream_name);
  

• 		snd_soc_dapm_device_event(socdev,
  

• 			SNDRV_CTL_POWER_D1);
  

• 	}
  

• codec_dai->pop_wait = 0;
  snd_soc_dapm_stream_event(codec,
  	codec_dai->playback.stream_name,
  

-- 1.5.3.5

---

Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel