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[alsa-devel] [PATCH] sound: lx_core: Few misc cleanups
by Maxime Ripard 17 Apr '14

17 Apr '14

This file had a few comments in french that are now in english, and took the opportunity to cleanup a bunch of #if 0 .. #endif and commented out code Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/pci/lx6464es/lx_core.c | 70 +++++++++++++++++--------------------------- 1 file changed, 27 insertions(+), 43 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 626ecad4dae7..002049770ff7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -22,8 +22,7 @@ * */ -/* #define RMH_DEBUG 1 */ - +#include <linux/bitops.h> #include <linux/module.h> #include <linux/pci.h> #include <linux/delay.h> @@ -496,15 +495,17 @@ int lx_dsp_es_check_pipeline(struct lx6464es *chip) int i; for (i = 0; i != CSES_TIMEOUT; ++i) { - /* - * le bit CSES_UPDATE_LDSV est Ã 1 dÃ©s que le macprog - * est pret. il re-passe Ã 0 lorsque le premier read a - * Ã©tÃ© fait. pour l'instant on retire le test car ce bit - * passe a 1 environ 200 Ã 400 ms aprÃ©s que le registre - * confES Ã Ã©tÃ© Ã©crit (kick du xilinx ES). + /* + * The bit CSES_UPDATE_LDSV is set to 1 as soon as the + * macprog is ready. He goes back to 0 whenever the + * first read has been issued. + * + * For now, we don't bother testing this bit because + * it's set to 1 between 200 to 400ms after the confES + * register has been written to (xilinx ES kick). * - * On ne teste que le bit CE. - * */ + * We only test the CE bit. + */ u32 cses = lx_dsp_reg_read(chip, eReg_CSES); @@ -602,8 +603,7 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, /* free */ *r_needed += 1; } - -#if 0 +#if DEBUG snd_printdd(LXP "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); for (i = 0; i < MAX_STREAM_BUFFER; ++i) { @@ -1049,9 +1049,9 @@ int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels, /* interrupt handling */ #define PCX_IRQ_NONE 0 -#define IRQCS_ACTIVE_PCIDB 0x00002000L /* Bit nÃÂ¸ 13 */ -#define IRQCS_ENABLE_PCIIRQ 0x00000100L /* Bit nÃÂ¸ 08 */ -#define IRQCS_ENABLE_PCIDB 0x00000200L /* Bit nÃÂ¸ 09 */ +#define IRQCS_ACTIVE_PCIDB BIT(13) +#define IRQCS_ENABLE_PCIIRQ BIT(8) +#define IRQCS_ENABLE_PCIDB BIT(9) static u32 lx_interrupt_test_ack(struct lx6464es *chip) { @@ -1108,25 +1108,21 @@ static int lx_interrupt_handle_async_events(struct lx6464es *chip, u32 irqsrc, int err; u32 stat[9]; /* answer from CMD_04_GET_EVENT */ - /* On peut optimiser pour ne pas lire les evenements vides - * les mots de rÃÂ©ponse sont dans l'ordre suivant : - * Stat[0] mot de status gÃÂ©nÃÂ©ral - * Stat[1] fin de buffer OUT pF - * Stat[2] fin de buffer OUT pf - * Stat[3] fin de buffer IN pF - * Stat[4] fin de buffer IN pf - * Stat[5] underrun poid fort - * Stat[6] underrun poid faible - * Stat[7] overrun poid fort - * Stat[8] overrun poid faible + /* We can optimize this to not read dumb events. + * Answer words are in the following order: + * Stat[0] general status + * Stat[1] end of buffer OUT pF + * Stat[2] end of buffer OUT pf + * Stat[3] end of buffer IN pF + * Stat[4] end of buffer IN pf + * Stat[5] MSB underrun + * Stat[6] LSB underrun + * Stat[7] MSB overrun + * Stat[8] LSB overrun * */ u64 orun_mask; u64 urun_mask; -#if 0 - int has_underrun = (irqsrc & MASK_SYS_STATUS_URUN) ? 1 : 0; - int has_overrun = (irqsrc & MASK_SYS_STATUS_ORUN) ? 1 : 0; -#endif int eb_pending_out = (irqsrc & MASK_SYS_STATUS_EOBO) ? 1 : 0; int eb_pending_in = (irqsrc & MASK_SYS_STATUS_EOBI) ? 1 : 0; @@ -1272,7 +1268,7 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) if (irqsrc & MASK_SYS_STATUS_CMD_DONE) goto exit; -#if 0 +#if DEBUG if (irqsrc & MASK_SYS_STATUS_EOBI) snd_printdd(LXP "interrupt: EOBI\n"); @@ -1310,18 +1306,6 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) "error during audio transfer\n"); } - if (async_escmd) { -#if 0 - /* backdoor for ethersound commands - * - * for now, we do not need this - * - * */ - - snd_printdd("lx6464es: interrupt requests escmd handling\n"); -#endif - } - exit: spin_unlock(&chip->lock); return IRQ_HANDLED; /* this device caused the interrupt */ -- 1.9.1

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Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Clemens Ladisch 17 Apr '14

17 Apr '14

mick wrote: > I tried the patch but it does not help. Still crackles. Do you get changing values, and any out-of-range values, when you execute this while it's playing? (while sleep 0.123; do grep 'Momentary freq' /proc/asound/card*/stream0; done) | uniq Regards, Clemens

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[alsa-devel] [PATCH] ASoC: rt5640: Add minimal support for RT5642
by Jarkko Nikula 17 Apr '14

17 Apr '14

We have been using rt5640.c codec driver with RT5642 codec chip before commit 022d21f004c1 ("ASoC: rt5640: add rt5639 support"). That commits starts using device ID reading in reset register for adding device specific controls and routes runtime. Now since device ID appears to be different between RT5640 and RT5642 the driver doesn't add those controls and routes that are valid also on RT5642. Fix this by adding a device ID found by debugging and minimal code for supporting RT5642. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/rt5640.c | 2 ++ sound/soc/codecs/rt5640.h | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b947c820697d..ffde5bf11789 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2000,6 +2000,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) switch (snd_soc_read(codec, RT5640_RESET)) { case RT5640_RESET_ID: + case RT5642_RESET_ID: snd_soc_add_codec_controls(codec, rt5640_specific_snd_controls, ARRAY_SIZE(rt5640_specific_snd_controls)); @@ -2150,6 +2151,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, { "rt5639", 0 }, + { "rt5642", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 3b50459a83b4..d31faeaa637e 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -16,6 +16,7 @@ #define RT5639_RESET_ID 0x0008 #define RT5640_RESET_ID 0x000c +#define RT5642_RESET_ID 0x0006 /* Info */ #define RT5640_RESET 0x00 -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl: Add explicit include of of.h
by Mark Brown 16 Apr '14

16 Apr '14

From: Mark Brown <broonie(a)linaro.org> Hopefully fixing a build failure reported by Stephen Rothwell - though quite why the other OF headers don't include this as well I'm not sure. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5428a1fda260..2fbbd8416c00 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -40,6 +40,7 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/spinlock.h> +#include <linux/of.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> -- 1.9.1

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[alsa-devel] [PATCH] of: add empty of_find_node_by_path() for !OF
by Alexander Shiyan 16 Apr '14

16 Apr '14

Add an empty version of of_find_node_by_path(). This fixes following build error for asoc tree: sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe': sound/soc/fsl/fsl_ssi.c:1471:2: error: implicit declaration of function 'of_find_node_by_path' [-Werror=implicit-function-declaration] sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- include/linux/of.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/linux/of.h b/include/linux/of.h index 919bf21..3bad8d1 100644 --- a/include/linux/of.h +++ b/include/linux/of.h @@ -374,6 +374,11 @@ static inline struct device_node *of_find_matching_node_and_match( return NULL; } +static inline struct device_node *of_find_node_by_path(const char *path) +{ + return NULL; +} + static inline struct device_node *of_get_parent(const struct device_node *node) { return NULL; -- 1.8.3.2

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[alsa-devel] [PATCH v2 linux-next] ALSA: pcm: 'BUG:' message unnecessarily triggers kerneloops
by Tim Gardner 16 Apr '14

16 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305480 The kerneloops-daemon scans dmesg for common crash signatures, among which is 'BUG:'. The message emitted by the PCM library is really a warning, so the most expedient thing to do seems to be to change the string. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: JongHo Kim <furmuwon(a)gmail.com> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- v2: Changed string preface from "OVER RUN" to "XRUN". As Takashi pointed out, 'Use a term "XRUN", as it's not always an overrun.' sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..9acc77e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "XRUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 1.9.1

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 16 Apr '14

16 Apr '14

>Could you send the output of 'lsusb -v', please? Here is a pastebin link to the output: -http://pastebin.com/nRULYjZh

1 0

[alsa-devel] [PATCH 00/18] ASoC: omap: Bind the omap-pcm platform driver to dai device
by Peter Ujfalusi 16 Apr '14

16 Apr '14

Hi, This should have been done a long time ago... Add support for binding the omap-pcm platform to a dai device. IN this way we can remove the 'omap-pcm-audio' platform device/driver need. The change will also going to simplify cases when we boot with DT and we can use the same node for cpu_dai and platform. Regards, Peter --- Peter Ujfalusi (18): ASoC: omap-pcm: Support for binding the platform driver to dai devices ASoC: omap-mcpdm: Assign the dai DMA data at earlier time ASoC: omap-mcpdm: Bind the platform driver to the dai driver when loading ASoC: omap-mcbsp: Assign the dai DMA data at earlier time ASoC: omap-mcbsp: Bind the platform driver to the dai driver when loading ASoC: omap-dmic: Assign the dai DMA data at earlier time ASoC: omap-dmic: Bind the platform driver to the dai driver when loading ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading ASoC: omap-abe-twl6040: Use the cpu_dai node to specify the platform driver ASoC: omap-twl4030: Use the same name/node for platform as the cpu_dai ASoC: am3517evm: Use the same name for platform as the cpu_dai ASoC: ams-delta: Use the same name for platform as the cpu_dai ASoC: n810: Use the same name for platform as the cpu_dai ASoC: omap3pandora: Use the same name for platform as the cpu_dai ASoC: osk5912: Use the same name for platform as the cpu_dai ASoC: rx51: Use the same name for platform as the cpu_dai ASoC: omap-hdmi-card: Use the same name for platform as the cpu_dai ASoC: omap-pcm: Drop the platform driver init code sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 8 ++------ sound/soc/omap/omap-dmic.c | 10 +++++++++- sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-hdmi.c | 7 ++++++- sound/soc/omap/omap-mcbsp.c | 19 ++++++++++++------- sound/soc/omap/omap-mcpdm.c | 15 +++++++++++---- sound/soc/omap/omap-pcm.c | 25 ++++++------------------- sound/soc/omap/omap-pcm.h | 35 +++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-twl4030.c | 10 ++++++++-- sound/soc/omap/omap3pandora.c | 4 ++-- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- 15 files changed, 97 insertions(+), 48 deletions(-) create mode 100644 sound/soc/omap/omap-pcm.h -- 1.9.2

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[alsa-devel] Using simple-card to replace kirkwood-t5325.c
by Andrew Lunn 16 Apr '14

16 Apr '14

Hi Folks I'm an ALSA newbie, so if i say anything stupid, please let me know. I'm trying to replace sound/soc/kirkwood/kirkwood-t5325.c with DT, using simple-card. I can get near, but i'm missing two things: There does not appear to be a way to represent this in DT: static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); return 0; } It appears that quite a few drivers need to enable pins. It also seems common to disable pins and to set them to NC. Is this a generic enough feature it could be added to the DT binding? The second thing i need is: static int t5325_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int freq; freq = params_rate(params) * 256; return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); } static struct snd_soc_ops t5325_ops = { .hw_params = t5325_hw_params, }; This seems a lot less common requirements. All the Marvell SoCs need it, but not many others. So i don't think it makes sense to add it directly to simple-card, otherwise simple-card quickly becomes complex-card as everybody else wants there quirks adding. Has there been any thoughts about turning simple-card.c into a library, or adding hooks so that a driver can modify the created dai_link structures to add in ops like this? Thanks Andrew

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 16 Apr '14

16 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

2 1

[alsa-devel] Add support for headset Mic of Dell
by Kailang 16 Apr '14

16 Apr '14

Hi Takashi, Add support for Dell machine for Headset Mic support. SSID: 0x0696 and 0x0698 Thanks. BR, Kailang

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[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Convert to module_platform_driver
by Alexander Stein 16 Apr '14

16 Apr '14

This reduces some boilerplate code. Signed-off-by: Alexander Stein <alexanders83(a)web.de> Acked-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/atmel/ac97c.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 05ec049..a04d2317 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1198,6 +1198,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) } static struct platform_driver atmel_ac97c_driver = { + .probe = atmel_ac97c_probe, .remove = atmel_ac97c_remove, .driver = { .name = "atmel_ac97c", @@ -1205,19 +1206,7 @@ static struct platform_driver atmel_ac97c_driver = { .pm = ATMEL_AC97C_PM_OPS, }, }; - -static int __init atmel_ac97c_init(void) -{ - return platform_driver_probe(&atmel_ac97c_driver, - atmel_ac97c_probe); -} -module_init(atmel_ac97c_init); - -static void __exit atmel_ac97c_exit(void) -{ - platform_driver_unregister(&atmel_ac97c_driver); -} -module_exit(atmel_ac97c_exit); +module_platform_driver(atmel_ac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); -- 1.9.2

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 16 Apr '14

16 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x1028067f), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7766fe..26895d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4621,6 +4621,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: cs42l51: split i2c from codec driver
by Brian Austin 16 Apr '14

16 Apr '14

This patch removes the i2c bus code from the codec driver and creates seperate i2c driver. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 +++- sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/cs42l51-i2c.c | 59 ++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l51.c | 77 +++++++++++----------------------------- sound/soc/codecs/cs42l51.h | 5 +++ 5 files changed, 91 insertions(+), 58 deletions(-) create mode 100644 sound/soc/codecs/cs42l51-i2c.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..5ee2eb5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC - select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C @@ -233,6 +233,10 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L51_I2C + tristate + select SND_SOC_CS42L51 + config SND_SOC_CS42L52 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..3046dc0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o @@ -152,6 +153,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c new file mode 100644 index 0000000..50d42ef --- /dev/null +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -0,0 +1,59 @@ +/* + * cs42l56.c -- CS42L51 ALSA SoC I2C audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <sound/soc.h> + +#include "cs42l51.h" + +static struct i2c_device_id cs42l51_i2c_id[] = { + {"cs42l51", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); + +static int cs42l51_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = cs42l51_regmap; + config.val_bits = 8; + config.reg_bits = 8; + + return cs42l51_probe(&i2c->dev, devm_regmap_init_i2c(i2c, &config)); +} + +static int cs42l51_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver cs42l51_i2c_driver = { + .driver = { + .name = "cs42l51", + .owner = THIS_MODULE, + }, + .probe = cs42l51_i2c_probe, + .remove = cs42l51_i2c_remove, + .id_table = cs42l51_i2c_id, +}; + +module_i2c_driver(cs42l51_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L51 I2C Driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 65fc37c..68ab2eb 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -29,7 +29,6 @@ #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/pcm.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include "cs42l51.h" @@ -484,7 +483,7 @@ static struct snd_soc_dai_driver cs42l51_dai = { .ops = &cs42l51_dai_ops, }; -static int cs42l51_probe(struct snd_soc_codec *codec) +static int cs42l51_codec_probe(struct snd_soc_codec *codec) { int ret, reg; @@ -511,7 +510,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { - .probe = cs42l51_probe, + .probe = cs42l51_codec_probe, .controls = cs42l51_snd_controls, .num_controls = ARRAY_SIZE(cs42l51_snd_controls), @@ -521,91 +520,55 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; -static const struct regmap_config cs42l51_regmap = { - .reg_bits = 8, - .val_bits = 8, - +const struct regmap_config cs42l51_regmap = { .max_register = CS42L51_CHARGE_FREQ, .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL_GPL(cs42l51_regmap); -static int cs42l51_i2c_probe(struct i2c_client *i2c_client, - const struct i2c_device_id *id) +int cs42l51_probe(struct device *dev, struct regmap *regmap) { struct cs42l51_private *cs42l51; - struct regmap *regmap; unsigned int val; int ret; - - regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap); - if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - dev_err(&i2c_client->dev, "Failed to create regmap: %d\n", - ret); - return ret; - } + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + cs42l51 = devm_kzalloc(dev, sizeof(struct cs42l51_private), + GFP_KERNEL); + if (!cs42l51) + return -ENOMEM; + + dev_set_drvdata(dev, cs42l51); /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { - dev_err(&i2c_client->dev, "failed to read I2C\n"); + dev_err(dev, "failed to read I2C\n"); goto error; } if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { - dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val); + dev_err(dev, "Invalid chip id: %x\n", val); ret = -ENODEV; goto error; } + dev_info(dev, "Cirrus Logic CS42L51, Revision: %02X\n", val & 0xFF); - dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", - val & 7); - - cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), - GFP_KERNEL); - if (!cs42l51) - return -ENOMEM; - - i2c_set_clientdata(i2c_client, cs42l51); - - ret = snd_soc_register_codec(&i2c_client->dev, + ret = snd_soc_register_codec(dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); error: return ret; } - -static int cs42l51_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id cs42l51_id[] = { - {"cs42l51", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs42l51_id); +EXPORT_SYMBOL_GPL(cs42l51_probe); static const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } }; MODULE_DEVICE_TABLE(of, cs42l51_of_match); - -static struct i2c_driver cs42l51_i2c_driver = { - .driver = { - .name = "cs42l51-codec", - .owner = THIS_MODULE, - .of_match_table = cs42l51_of_match, - }, - .id_table = cs42l51_id, - .probe = cs42l51_i2c_probe, - .remove = cs42l51_i2c_remove, -}; - -module_i2c_driver(cs42l51_i2c_driver); - MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 2beeb17..641ef18 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -18,6 +18,11 @@ #ifndef _CS42L51_H #define _CS42L51_H +struct device; + +extern const struct regmap_config cs42l51_regmap; +int cs42l51_probe(struct device *dev, struct regmap *regmap); + #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 #define CS42L51_CHIP_REV_B 0x01 -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: Fix snd_soc_kcontrol_platform() return type
by Lars-Peter Clausen 16 Apr '14

16 Apr '14

This should obviously be snd_soc_platform * and not snd_soc_codec * Fixes: f6272ff8a5f4 ("ASoC: Add snd_soc_kcontrol_platform() helper function") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3396594..488e749 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1252,7 +1252,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( * registered with snd_soc_add_platform_controls() or via table based setup of * a snd_soc_platform_driver. Otherwise the behavior is undefined. */ -static inline struct snd_soc_codec *snd_soc_kcontrol_platform( +static inline struct snd_soc_platform *snd_soc_kcontrol_platform( struct snd_kcontrol *kcontrol) { return snd_kcontrol_chip(kcontrol); -- 1.8.0

2 1

[alsa-devel] [PATCH 0/5] DT support for N900 soundcard (rx51-audio)
by Sebastian Reichel 15 Apr '14

15 Apr '14

Hi, This patchset adds DT support in rx51-audio. I tested it on my Nokia N900 and was able to play a wav file using aplay. I have not tested the whole functionality, but output via speakers, headphones and the related enable controls seem to work. The patches can be applied cleanly to torvalds/linux.git:master and broonie/sound.git:for-next. -- Sebastian Pali Rohár (1): ASoC: omap: rx51: Use devm_snd_soc_register_card Sebastian Reichel (4): ASoC: Allow Aux Codecs to be specified using DT ASoC: RX-51: Add DT support to sound driver ASoC: RX-51: Convert to table based DAPM setup ASoC: tlv320aic3x: fix shared reset pin for DT .../devicetree/bindings/sound/nokia,rx51.txt | 27 +++ include/sound/rx51.h | 18 ++ include/sound/soc.h | 13 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 248 +++++++++++++-------- sound/soc/soc-core.c | 68 +++--- 8 files changed, 266 insertions(+), 124 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt create mode 100644 include/sound/rx51.h -- 1.9.1

3 11

[alsa-devel] [PATCH] ASoC: alc5623: Fix regmap endianness
by Andrew Lunn 15 Apr '14

15 Apr '14

Commit 0cd257bf9b9b0cbb4fa1a5c988a232506997867c, "ASoC: alc5623: Convert to direct regmap API usage" broke probing of the codec, because of wrong endinness of the ID and codec version read from the device. Fix this by removing the existing flipping of the endiannes, and extracting the codec type byte from the word from the regmap. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> --- sound/soc/codecs/alc5623.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index f500905e9373..2acf82f4a08a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1018,13 +1018,13 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); return ret; } - vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); if (ret < 0) { dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); return ret; } + vid2 >>= 8; if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { dev_err(&client->dev, "unknown or wrong codec\n"); -- 1.9.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix a self assignment in sst_mem_block_alloc_scratch()
by Christian Engelmayer 15 Apr '14

15 Apr '14

Remove a self assignment in sst_mem_block_alloc_scratch(). When calculating buffer sizes there is no need for statements without effect. Detected by Coverity: CID 1195249. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- sound/soc/intel/sst-firmware.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f768710..f24619a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -505,9 +505,7 @@ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp) /* calculate required scratch size */ list_for_each_entry(sst_module, &dsp->module_list, list) { - if (scratch->s.size > sst_module->s.size) - scratch->s.size = scratch->s.size; - else + if (scratch->s.size < sst_module->s.size) scratch->s.size = sst_module->s.size; } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: wm8997: Fix compile error
by Lars-Peter Clausen 15 Apr '14

15 Apr '14

The macro's name is SOC_ENUM, not SOC_VALUE. Fixes: e13dd8ce ("ASoC: wm8997: Replace usage deprecated MUX/ENUM macros") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I'm getting to old for this... --- sound/soc/codecs/wm8997.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 61926cb..bb9b47b 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -245,8 +245,8 @@ SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), -SOC_VALUE("ISRC1 FSL", arizona_isrc_fsl[0]), -SOC_VALUE("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -286,8 +286,8 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_VALUE("HPOUT1 OSR", wm8997_hpout_osr[0]), -SOC_VALUE("EPOUT OSR", wm8997_hpout_osr[1]), +SOC_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix incorrect condition check in trigger()
by Nicolin Chen 15 Apr '14

15 Apr '14

Patch ASoC: fsl_sai: Fix buggy configurations in trigger() doesn't entirely fix the condition: FRDE of the current substream direction is being cleared while the code is still using the non-updated one. Thus this patch fixes this issue by checking the opposite one's FRDE alone since the current one's is absolutely disabled. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9858d42..1716a41 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -343,7 +343,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_xIE_MASK, 0); - if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + /* Check if the opposite FRDE is also disabled */ + if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { regmap_update_bits(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_TERE, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, -- 1.8.4

2 2

[alsa-devel] [PATCH v2 0/2] ASoC: fsl_sai: Add bus clock and mclks with clock controls
by Nicolin Chen 15 Apr '14

15 Apr '14

This series of patches add clock controls to fsl_sai driver and updates the vf610.dtsi accordingly. Changelog v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. Nicolin Chen (2): ASoC: fsl_sai: Add clock controls for SAI ARM: dts: Append clock bindings for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 8 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 50 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 59 insertions(+), 9 deletions(-) -- 1.8.4

4 9

[alsa-devel] [PATCH RFC v2 0/2] Fix simple-card *-master DT parameter handling
by Jyri Sarha 15 Apr '14

15 Apr '14

Since RFC: - fixed commit msg typo - added include/sound/soc.h changes too The sematics of bitclock-master and frame-master DT parameters should depend on whether they are found from a cpu-dai or codec sub-node. - bitclock-master in cpu-dai node means Codec-Bitclock-Slave - frame-master in cpu-dai node means Codec-Frame-Slave - bitclock-master in codec node means Codec-Bitclock-Master - frame-master in codec node means Codec-Frame-Master For example in a cpu-dai mode bitclock-master parameter should produce SND_SOC_DAIFMT_CBS_* daifmt flags and a codec node SND_SOC_DAIFMT_CBM_* flags. Best regards, Jyri Jyri Sarha (2): ASoC: core: Add is_cpu_dai_node-parameter to snd_soc_of_parse_daifmt() ASoC: simple-card: Take snd_soc_of_parse_daifmt() change in to account include/sound/soc.h | 3 ++- sound/soc/generic/simple-card.c | 13 ++++++++----- sound/soc/soc-core.c | 8 +++++++- 3 files changed, 17 insertions(+), 7 deletions(-) -- 1.7.9.5

4 12

[alsa-devel] [PATCH] ASoC: fsl_sai: Use FSL_SAI_xXR() and regmap_update_bits() to simplify code
by Nicolin Chen 14 Apr '14

14 Apr '14

By doing this, the driver can drop around 50 lines and become neater. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 101 ++++++++++++------------------------------------ 1 file changed, 25 insertions(+), 76 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5fff2e1..9858d42 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -93,16 +93,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, reg_cr2; - - if (fsl_dir == FSL_FMT_TRANSMITTER) - reg_cr2 = FSL_SAI_TCR2; - else - reg_cr2 = FSL_SAI_RCR2; - - regmap_read(sai->regmap, reg_cr2, &val_cr2); - - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + bool tx = fsl_dir == FSL_FMT_TRANSMITTER; + u32 val_cr2 = 0; switch (clk_id) { case FSL_SAI_CLK_BUS: @@ -121,7 +113,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx), + FSL_SAI_CR2_MSEL_MASK, val_cr2); return 0; } @@ -153,22 +146,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, val_cr4, reg_cr2, reg_cr4; - - if (fsl_dir == FSL_FMT_TRANSMITTER) { - reg_cr2 = FSL_SAI_TCR2; - reg_cr4 = FSL_SAI_TCR4; - } else { - reg_cr2 = FSL_SAI_RCR2; - reg_cr4 = FSL_SAI_RCR4; - } + bool tx = fsl_dir == FSL_FMT_TRANSMITTER; + u32 val_cr2 = 0, val_cr4 = 0; - regmap_read(sai->regmap, reg_cr2, &val_cr2); - regmap_read(sai->regmap, reg_cr4, &val_cr4); - - if (sai->big_endian_data) - val_cr4 &= ~FSL_SAI_CR4_MF; - else + if (!sai->big_endian_data) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -189,7 +170,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * frame sync asserts with the first bit of the frame. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: /* @@ -199,7 +179,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * data word. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; break; @@ -209,7 +188,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * frame sync asserts with the first bit of the frame. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; case SND_SOC_DAIFMT_RIGHT_J: @@ -247,23 +225,22 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: - val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; - val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; - val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: - val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; default: return -EINVAL; } - regmap_write(sai->regmap, reg_cr2, val_cr2); - regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx), + FSL_SAI_CR2_BCP | FSL_SAI_CR2_BCD_MSTR, val_cr2); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx), + FSL_SAI_CR4_MF | FSL_SAI_CR4_FSE | + FSL_SAI_CR4_FSP | FSL_SAI_CR4_FSD_MSTR, val_cr4); return 0; } @@ -290,29 +267,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - reg_cr4 = FSL_SAI_TCR4; - reg_cr5 = FSL_SAI_TCR5; - reg_mr = FSL_SAI_TMR; - } else { - reg_cr4 = FSL_SAI_RCR4; - reg_cr5 = FSL_SAI_RCR5; - reg_mr = FSL_SAI_RMR; - } - - regmap_read(sai->regmap, reg_cr4, &val_cr4); - regmap_read(sai->regmap, reg_cr4, &val_cr5); - - val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; - val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - - val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; - val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; - val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; + u32 val_cr4 = 0, val_cr5 = 0; if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -320,18 +278,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); - val_mr = ~0UL - ((1 << channels) - 1); - regmap_write(sai->regmap, reg_cr4, val_cr4); - regmap_write(sai->regmap, reg_cr5, val_cr5); - regmap_write(sai->regmap, reg_mr, val_mr); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx), + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + val_cr4); + regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx), + FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | + FSL_SAI_CR5_FBT_MASK, val_cr5); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1)); return 0; } @@ -401,8 +361,8 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct device *dev = &sai->pdev->dev; - u32 reg; int ret; ret = clk_prepare_enable(sai->bus_clk); @@ -411,12 +371,7 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, return ret; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg = FSL_SAI_TCR3; - else - reg = FSL_SAI_RCR3; - - regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, FSL_SAI_CR3_TRCE); return 0; @@ -426,15 +381,9 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg = FSL_SAI_TCR3; - else - reg = FSL_SAI_RCR3; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, - ~FSL_SAI_CR3_TRCE); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0); clk_disable_unprepare(sai->bus_clk); } -- 1.8.4

2 1

[alsa-devel] [PATCHv2 0/4] Add ESAI & SPDIF Vybrid and LS1+ support.
by Xiubo Li 14 Apr '14

14 Apr '14

Changes in V2: - Revise some issue from Nicolin Chen and Fabio Estevam. Xiubo Li (4): ASoC: esai: use the precise definition of 'ret'. ASoC: spdif: Sort the header files alphabetically. ASoC: esai: Add VF610+ compatibles support. ASoC: spdif: Add VF610+ compatibles support. sound/soc/fsl/fsl_esai.c | 9 ++++++--- sound/soc/fsl/fsl_spdif.c | 9 +++++---- 2 files changed, 11 insertions(+), 7 deletions(-) -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: dapm: don't skip evaluate the power of widget with force set
by Xiang Xiao 14 Apr '14

14 Apr '14

Signed-off-by: Xiang Xiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..14af77e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1673,7 +1673,7 @@ static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, /* If the peer is already in the state we're moving to then we * won't have an impact on it. */ - if (power != peer->power) + if (peer->force || power != peer->power) dapm_mark_dirty(peer, "peer state change"); } @@ -1683,7 +1683,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, { struct snd_soc_dapm_path *path; - if (w->power == power) + if (!w->force && w->power == power) return; trace_snd_soc_dapm_widget_power(w, power); -- 1.8.1.2

2 1

[alsa-devel] [RFT v2 0/6] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 14 Apr '14

14 Apr '14

Hi Mark and Liam, Here is the second version of the multicodec series. I mostly rebased the code to asoc-next, added smaller patches to introduce helpers before the main patch and thus reduce the overal size of that patch and modified the TDM slot fixup. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test the series on various setup to ensure this is not generating any regression. Hence the RFT for this version. The series is based on asoc-v3.15-2-245-g876dea8. Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: - Split the first version in several patches. - Remove the TDM fixup --- Misael Lopez Cruz (6): ASoC: core: Add helpers for codec and codec_dai search ASoC: core: Add helpers for codec DAI probe & remove ASoC: core: Add helper for DAI widgets linking ASoC: core: Add function for ac97 codec registration ASoC: core: Add helpers for dai link and aux dev init ASoC: core: Add support for DAI multicodec include/sound/soc-dai.h | 5 + include/sound/soc.h | 16 ++ sound/soc/soc-core.c | 578 ++++++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 39 +++- sound/soc/soc-pcm.c | 438 +++++++++++++++++++++++++----------- 5 files changed, 761 insertions(+), 315 deletions(-) -- 1.8.3.2

3 11

[alsa-devel] [PATCH] ASoC: jack: export gpio detect
by cfreeman＠nvidia.com 14 Apr '14

14 Apr '14

From: Christopher Freeman <cfreeman(a)nvidia.com> Export the gpio detect function so machine drivers may call it. Interrupts for the jack may be disabled during sleep, so this allows a machine driver to have the jack status updated during resume. Signed-off-by: Christopher Freeman <cfreeman(a)nvidia.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..36383cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -442,6 +442,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio); #endif /* codec register bit access */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..fd97cc5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -231,7 +231,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); #ifdef CONFIG_GPIOLIB /* gpio detect */ -static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; @@ -251,6 +251,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) snd_soc_jack_report(jack, report, gpio->report); } +EXPORT_SYMBOL_GPL(snd_soc_jack_gpio_detect); /* irq handler for gpio pin */ static irqreturn_t gpio_handler(int irq, void *data) -- 1.8.3.2 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

3 5

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not touch 0x04 register above McASP_VERSION_2
by Peter Ujfalusi 14 Apr '14

14 Apr '14

This register is not defined in TI81xx and on AM335x/AM437x it is the SYSCONFIG register which should not be touched by drivers since it is related to PM and handled by the generic PM code. This register write was there since the first time the davinci-mcasp driver was appeared in the kernel. The reason why it did not caused any issues on AM335x/AM437x is that it sets bit 1 in SYSCONFIG register which in turn will enable the smart-idle mode. This is the default mode and this is the mode McASP should be in also when in use. On TI81xx the register is not defined. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ce0e55a4304c..50a824d9dbb7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -495,7 +495,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int active_serializers, numevt, n; u32 reg; /* Default configuration */ - if (mcasp->version != MCASP_VERSION_4) + if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ -- 1.9.1

2 1

[alsa-devel] [PATCH 00/17] Removed deprecated ENUM/MUX macros
by Lars-Peter Clausen 14 Apr '14

14 Apr '14

Hi, This is a follow-up to the enum consolidation series. It updates all users of the now redundant and deprecated macros and once that is done removes the macros. - Lars Lars-Peter Clausen (17): ASoC: adau1373: Replace usage deprecated MUX/ENUM macros ASoC: adav80x: Replace usage deprecated MUX/ENUM macros ASoC: arizona: Replace usage deprecated MUX/ENUM macros ASoC: max98090: Replace usage deprecated MUX/ENUM macros ASoC: mc13783: Replace usage deprecated MUX/ENUM macros ASoC: pcm512x: Replace usage deprecated SOC_VALUE_ENUM macro ASoC: rt5640: Replace usage deprecated MUX/ENUM macros ASoC: wm2200: Replace usage deprecated MUX/ENUM macros ASoC: wm5100: Replace usage deprecated MUX/ENUM macros ASoC: wm5102: Replace usage deprecated MUX/ENUM macros ASoC: wm5110: Replace usage deprecated MUX/ENUM macros ASoC: wm8988: Replace usage deprecated MUX/ENUM macros ASoC: wm5102: Replace usage deprecated SOC_VALUE_ENUM macro ASoC: wm8994: Replace usage deprecated MUX/ENUM macros ASoC: wm8995: Replace usage deprecated MUX/ENUM macros ASoC: wm8997: Replace usage deprecated MUX/ENUM macros ASoC: Remove deprecated ENUM/MUX macros include/sound/soc-dapm.h | 12 ------------ include/sound/soc.h | 2 -- sound/soc/codecs/adau1373.c | 7 +++---- sound/soc/codecs/adav80x.c | 8 ++++---- sound/soc/codecs/arizona.h | 4 ++-- sound/soc/codecs/max98090.c | 5 ++--- sound/soc/codecs/mc13783.c | 8 ++++---- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt5640.c | 4 ++-- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 4 ++-- sound/soc/codecs/wm5102.c | 14 +++++++------- sound/soc/codecs/wm5110.c | 18 +++++++++--------- sound/soc/codecs/wm8988.c | 8 ++++---- sound/soc/codecs/wm8994.c | 12 ++++++------ sound/soc/codecs/wm8995.c | 10 ++++------ sound/soc/codecs/wm8997.c | 12 ++++++------ 17 files changed, 58 insertions(+), 76 deletions(-) -- 1.8.0

2 18

[alsa-devel] [PATCH 1/6] ASoC: pcm512x: Use CONFIG_PM_RUNTIME macro
by Sachin Kamat 14 Apr '14

14 Apr '14

Fixes the following compilation warnings: sound/soc/codecs/pcm512x.c:520:12: warning: ‘pcm512x_suspend’ defined but not used [-Wunused-function] sound/soc/codecs/pcm512x.c:545:12: warning: ‘pcm512x_resume’ defined but not used [-Wunused-function] Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4b4c0c7..5112420 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -517,6 +517,7 @@ void pcm512x_remove(struct device *dev) } EXPORT_SYMBOL_GPL(pcm512x_remove); +#ifdef CONFIG_PM_RUNTIME static int pcm512x_suspend(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -578,6 +579,7 @@ static int pcm512x_resume(struct device *dev) return 0; } +#endif const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) -- 1.7.9.5

2 11

[alsa-devel] [PATCH 0/7] ASoC:davinci-mcasp: bc polarity fix and proper format support
by Peter Ujfalusi 14 Apr '14

14 Apr '14

Hi, The bit clock polarity has been configured incorrectly in the McASP driver for a long time. IB_NF, NB_IF and IB_IF was not correct on the receive side since they were selecting the same edge as we used for the tx. The driver only had support for DSP_B mode (and probably AC97, but I can not test that). All other formats were broken (DSP_A, I2S, LEFT_J, etc). With this series the following formats will be supported by the McASP driver: - DSP_B - AC97 - DSP_A (new) - I2S (new) - LEFT_J (new) Non supported modes will return error from now on. Regards, Peter --- Peter Ujfalusi (7): ASoC: davinci-mcasp: Fix bit clock polarity settings ASoC: davinci-mcasp: Format data delay configuration enhancement ASoC: davinci-mcasp: Support for DSP_A format ASoC: davinci-mcasp: Move the FS polarity change out from the switch case ASoC: davinci-mcasp: Add support for I2S format ASoC: davinci-mcasp: Support for LEFT_J format ASoC: davinci-mcasp: Remove excess empty lines from davinci_mcasp_set_dai_fmt() sound/soc/davinci/davinci-mcasp.c | 76 +++++++++++++++++++++++++-------------- 1 file changed, 50 insertions(+), 26 deletions(-) -- 1.9.1

3 13

[alsa-devel] iMX233 audio capture: glitches on task switch?
by Gilles Grégoire 14 Apr '14

14 Apr '14

Dear alsa developers, I need help debugging a driver for the builtin codec of Freescale's iMX233 SoC. I am facing an issue when capturing audio: garbage data (glitches) appears periodically in the recorded audio data. Audio playback looks fine, as far as I can tell. I am using kernel 3.13.6. A driver by Freescale exists for a 2.6.35 kernel but it was never submitted to mainline. Therefore, the driver I use is a port of Freescale's driver to kernel 3.13.6 by Michal Ulianko. You can find it in the attached patch+dts. Please note that this is still a work in progress. The driver is actually quite simple: it simply sets up the codec/cpu DAI and uses the generic dma engine through snd_dmaengine_pcm_register() to make the audio run. I am using arecord to capture the audio from the default input (the mic), which is _not_ connected in my setup. The command line I am using is the following: $ arecord --disable-format --disable-resample --fatal-errors --disable-channels --buffer-size=32768 --period-size=4096 -r 22050 -f S16_LE -c 2 -d 1 /tmp/test.wav What happens: the recorded audio data contains glitches at periodic intervals which contain garbage data. No underrun/overrun occurs during capture. I suspect that the problem is not in the driver's code itself, but could be the symptom of a bug somewhere else (possibly in mxs dma code), because of the following observations I did: * there are 2 kind of glitches: short (~3 samples) and long (~25 samples) depending on the kernel configuration. Both kinds may exists on the same capture. * short glitches exist only with CONFIG_HZ_PERIODIC=y. The period of those glitches is _exactly_ 4ms, which corresponds to the HZ value in my kernel config (250Hz). * long glitches are always there. The period of those glitches is directly related to the --period-bytes parameter of arecord: halving the --period-bytes parameter halves the period of the glitches. You can find attached a screenshot showing the glitches for the "dynticks" case (where only long glitches exist) and for the "periodic timer" case (where both kinds are visible). These observations lead me to believe that the glitches are related to task switching. Here is my best guess: * the short glitches come from the HZ tick invoking the scheduler. * the long glitches come from the kernel code waking up the arecord process when enough audio data was received. I spent a long time trying to narrow down the problem with no success. My main problem is that I have no idea what kind of bug could cause these symptoms. This is where I need help: I would welcome any advice on what to look for. Best regards, -- Gilles

2 2

[alsa-devel] snd-usb-audio issues??
by Richard Storer 14 Apr '14

14 Apr '14

Hi, this is my first time post on the ML so apologies if my message is not to the required standard. I am currently using a Raspberry Pi ARM computer it's running a customised version of Debian called Raspbian: Linux raspberrypi 3.10.36+ #665 PREEMPT Wed Apr 9 15:01:53 BST 2014 armv6l GNU/Linux For audio playback and capture I am using a Creative Sound Blaster Live 24bit external: ID 041e:3040 Creative Technology, Ltd SoundBlaster Live! 24-bit External SB0490 I am currently experiencing 2 problems with this device which I believe are related to the snd-usb-audio driver. My first issue is for some reason booting Linux with my device already attached to the USB port produces no audio output, however capture is functioning. I've tried mute, unmute via alsamixer but it seems like the amplifier circuit inside hasn't turned itself on properly. The cure for this (and 100% reproducible) would be to unplug / replug the USB cable or simply connect to USB after Linux has fully loaded. I don't understand why this should be the case surely it doesn't matter at what stage the device is connected?? My next issue I feel is a little bit more serious and from what I understand affects all versions of Linux even the x86 distros. I use my Sound Blaster to capture stereo audio via Line-In which I then stream in realtime via Darkice. While Line-In capture does function on this device the signal levels are extremely low. Unfortunately there is no Line-In mixer control for my device the only capture control I have is Mic and that doesn't control Line-In. I have search for many months for a solution to the lack of Line-In control but I have come to the conclusion that something is broken in the driver or it got missed out. Is anyone aware of these issues or could at least offer a solution to increase the Line-In mixer levels maybe communicate with the USB device directly without ALSA's assistance?? Richard S.

3 8

[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

3 2

[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

3 2

[alsa-devel] Add support of alc288
by Kailang 14 Apr '14

14 Apr '14

Hi Takashi, We have new codec release for ALC288. Thanks. BR, Kailang

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fix clock prepare/unprepare
by Kuninori Morimoto 14 Apr '14

14 Apr '14

From: Ben Dooks <ben.dooks(a)codethink.co.uk> As with the previous commit, before a clock can be used it must be prepared for use. Change from clk_enable() and clk_disable() to the versions of the calls which also prepare and un-prepare the clocks. Will fix warnings from the clock code when this is used. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch is for current linus/master tree sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6232b7d..4d0720e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -258,7 +258,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_enable(src->clk); + clk_prepare_enable(src->clk); return 0; } @@ -269,7 +269,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_disable(src->clk); + clk_disable_unprepare(src->clk); return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4b7e206..1d8387c 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -171,7 +171,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, u32 cr; if (0 == ssi->usrcnt) { - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); if (rsnd_dai_is_clk_master(rdai)) { if (rsnd_ssi_clk_from_parent(ssi)) @@ -230,7 +230,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, rsnd_ssi_master_clk_stop(ssi); } - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); } dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); -- 1.7.9.5

2 3

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 14 Apr '14

14 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

1 0

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 13 Apr '14

13 Apr '14

Continued from here: http://article.gmane.org/gmane.linux.alsa.devel/121575 > http://www.alsa-project.org/main/index.php/XRUN_Debug > > # Enable basic debugging and dump stack, check hardware pointer on the > period update > # Usefull to just see, if PCM stream is stopped for a reason (usually wrong > audio process timing from scheduler) > # And to check the values from driver > > echo 11 > /proc/asound/card0/pcm0p/xrun_debug Okay, I not compiled my own kernel with CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, CONFIG_SND_DEBUG all enabled in the kernel configuration, I have an xrun_debug file, but it doesn't seem to list anything, then I also tried with cat instead of echo, and just a 0 showed up: ___________________________________________________________________ $ echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied # echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied $ cat /proc/asound/card0/pcm0p/xrun_debug 0 ___________________________________________________________________

2 3

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 12 Apr '14

12 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

1 0

[alsa-devel] [PATCH] ALSA: pcm: 'BUG:' message unnecessarily triggers kerneloops
by Tim Gardner 11 Apr '14

11 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305480 The kerneloops-daemon scans dmesg for common crash signatures, among which is 'BUG:'. The message emitted by the PCM library is really a warning, so the most expedient thing to do seems to be to change the string. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: JongHo Kim <furmuwon(a)gmail.com> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..2a0563d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "OVER RUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 11 Apr '14

11 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changelog v6: * Dropped the dtsi change since it's no more demanded with this patch. * Removed Shawn's ACK since we no long change dtsi in this case. v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 1

[alsa-devel] Broken alc562x codec in linus/master
by Andrew Lunn 10 Apr '14

10 Apr '14

Hi Mark I found a few issues with the changes you have made to sound/soc/codecs/alc5623.c which is currently in linus master branch. The patch causing the trouble seems to be: commit 0cd257bf9b9b0cbb4fa1a5c988a232506997867c Author: Mark Brown <broonie(a)linaro.org> Date: Thu Feb 20 09:04:06 2014 +0900 ASoC: alc5623: Convert to direct regmap API usage Convert to directly use the regmap API, allowing us to eliminate the last user of the ASoC level I/O implementations (there are still open coded I/O implementations in drivers), avoiding duplicating code in regmap. We no longer cache the entire CODEC register map on probe since the more advanced cache infrastructure in regmap is able to fill the cache on demand. Signed-off-by: Mark Brown <broonie(a)linaro.org> At boot i see: alc562x-codec 0-001a: unknown or wrong codec alc562x-codec 0-001a: Expected 10ec:21, got ec10:2103 There are two issues here: - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); It looks like these are returning different endianness. This explains the 10ec vs ec10. Can the regmap flip the endianness? The second is: - vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); + return ret; It was reading a byte, but now it seems to get a word. Hence 21 vs 2103. Once the endianness is sorted out, we can mask for the lower byte. Thanks Andrew

1 0

[alsa-devel] [PATCH v6] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changelog v6: * Dropped the dtsi change since it's no more demanded with this patch. * Removed Shawn's ACK since we no long change dtsi in this case. v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

1 0

[alsa-devel] [PATCH v5] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver, meanwhile, corrects the existing DTS accordingly so those platforms can benifit from the further feature with different clock sources. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changelog v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- arch/arm/boot/dts/vf610.dtsi | 6 ++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 4 files changed, 50 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi index d31ce1b..4c3cd59 100644 --- a/arch/arm/boot/dts/vf610.dtsi +++ b/arch/arm/boot/dts/vf610.dtsi @@ -139,8 +139,10 @@ compatible = "fsl,vf610-sai"; reg = <0x40031000 0x1000>; interrupts = <0 86 0x04>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; status = "disabled"; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: rt5640: Add the string "rt5639" to the list of I2C device IDs
by Oder Chiou 10 Apr '14

10 Apr '14

The patch adds the string "rt5639" to the list of I2C device IDs. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index d672f44..2e7132b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1,5 +1,5 @@ /* - * rt5640.c -- RT5640 ALSA SoC audio codec driver + * rt5640.c -- RT5640/RT5639 ALSA SoC audio codec driver * * Copyright 2011 Realtek Semiconductor Corp. * Author: Johnny Hsu <johnnyhsu(a)realtek.com> @@ -2164,6 +2164,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, + { "rt5639", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); @@ -2322,6 +2323,6 @@ static struct i2c_driver rt5640_i2c_driver = { }; module_i2c_driver(rt5640_i2c_driver); -MODULE_DESCRIPTION("ASoC RT5640 driver"); +MODULE_DESCRIPTION("ASoC RT5640/RT5639 driver"); MODULE_AUTHOR("Johnny Hsu <johnnyhsu(a)realtek.com>"); MODULE_LICENSE("GPL v2"); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v4] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver, meanwhile, corrects the existing DTS accordingly so those platforms can benifit from the further feature with different clock sources. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> CC: Xiubo Li <Li.Xiubo(a)freescale.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- @Xiubo Even though you've tested it before, I'd still like to wait for your test result to this newer version. Changelog v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 51 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 61 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi index d31ce1b..4c3cd59 100644 --- a/arch/arm/boot/dts/vf610.dtsi +++ b/arch/arm/boot/dts/vf610.dtsi @@ -139,8 +139,10 @@ compatible = "fsl,vf610-sai"; reg = <0x40031000 0x1000>; interrupts = <0 86 0x04>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; status = "disabled"; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..7cd4af9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,23 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; + struct device *dev = &sai->pdev->dev; + u32 reg, i; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock\n"); + return ret; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + ret = clk_prepare_enable(sai->mclk_clk[i]); + if (ret) { + dev_err(dev, "failed to enable mclk%d clock\n", i + 1); + goto err; + } + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -412,13 +428,20 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE); return 0; + +err: + for (; i > 0; i--) + clk_disable_unprepare(sai->mclk_clk[i - 1]); + clk_disable_unprepare(sai->bus_clk); + + return ret; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; + u32 reg, i; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +450,10 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) + clk_disable_unprepare(sai->mclk_clk[i]); + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +586,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +610,27 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock\n"); + return PTR_ERR(sai->bus_clk); + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock\n", i + 1); + return PTR_ERR(sai->mclk_clk[i]); + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: core: Fix Sparse incompatible types warning
by Krzysztof Kozlowski 10 Apr '14

10 Apr '14

Fix following Sparse warning: sound/soc/soc-core.c:252:20: error: incompatible types in comparison expression (different type sizes) Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 051c006281f5..41f6178249df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -249,7 +249,7 @@ static ssize_t codec_reg_write_file(struct file *file, struct snd_soc_codec *codec = file->private_data; int ret; - buf_size = min(count, (sizeof(buf)-1)); + buf_size = min(count, (size_t)(sizeof(buf)-1)); if (copy_from_user(buf, user_buf, buf_size)) return -EFAULT; buf[buf_size] = 0; -- 1.8.3.2

4 7

[alsa-devel] [PATCH 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 10 Apr '14

10 Apr '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of thw module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../bindings/sound/nvidia,tegra124-hda.txt | 20 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 695 +++++++++++++++++++++ 4 files changed, 731 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt new file mode 100644 index 0000000..c9256d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt @@ -0,0 +1,20 @@ +NVIDIA Tegra124 HDA controller + +Required properties: +- compatible : "nvidia,tegra124-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the hda controller. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda"; + reg = <0x0 0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..4f6ee6b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,20 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "Tegra HD Audio" + select SND_HDA + help + Say Y here to support the HDA controller present in Nvidia + Tegra SoCs + + This options enables support for the HD Audio controller + present in some Nvidia Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..c36bbca --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,695 @@ +/* + * + * Implementation of primary alsa driver code base for Tegra HDA. + * + * 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. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/mutex.h> +#include <linux/reboot.h> +#include <linux/io.h> +#include <linux/pm_runtime.h> +#include <linux/of_device.h> +#include <linux/time.h> +#include <linux/completion.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <linux/firmware.h> +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +#define DRV_NAME "tegra-hda" + +/* Defines for Nvidia Tegra HDA support */ +#define NVIDIA_TEGRA_HDA_BAR0_OFFSET 0x8000 + +#define NVIDIA_TEGRA_HDA_CFG_CMD_OFFSET 0x1004 +#define NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET 0x1010 + +#define NVIDIA_TEGRA_HDA_ENABLE_IO_SPACE (1 << 0) +#define NVIDIA_TEGRA_HDA_ENABLE_MEM_SPACE (1 << 1) +#define NVIDIA_TEGRA_HDA_ENABLE_BUS_MASTER (1 << 2) +#define NVIDIA_TEGRA_HDA_ENABLE_SERR (1 << 8) +#define NVIDIA_TEGRA_HDA_DISABLE_INTR (1 << 10) +#define NVIDIA_TEGRA_HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define NVIDIA_TEGRA_HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define NVIDIA_TEGRA_HDA_IPFS_CONFIG 0x180 +#define NVIDIA_TEGRA_HDA_IPFS_EN_FPCI 0x1 + +#define NVIDIA_TEGRA_HDA_IPFS_FPCI_BAR0 0x80 +#define NVIDIA_TEGRA_HDA_FPCI_BAR0_START 0x40 + +#define NVIDIA_TEGRA_HDA_IPFS_INTR_MASK 0x188 +#define NVIDIA_TEGRA_HDA_IPFS_EN_INTR (1 << 16) + +struct hda_tegra_data { + struct azx chip; + struct platform_device *pdev; + struct clk **platform_clks; + int platform_clk_count; + void __iomem *remap_config_addr; +}; + +static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; +module_param_array(probe_mask, int, NULL, 0444); +MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int *power_save_addr = &power_save; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, + int type, + size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, + chip->card->dev, + size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +#define MASK_LONG_ALIGN 0x3UL +#define SHIFT_BYTE 3 +#define SHIFT_BITS(addr) \ + (((unsigned int)(addr) & MASK_LONG_ALIGN) << SHIFT_BYTE) +#define ADDR_ALIGN_L(addr) \ + (void *)((unsigned int)(addr) & ~MASK_LONG_ALIGN) +#define MASK(bits) (BIT(bits) - 1) +#define MASK_REG(addr, bits) (MASK(bits) << SHIFT_BITS(addr)) + +static void tegra_hda_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 tegra_hda_readl(u32 *addr) +{ + return readl(addr); +} + +static void tegra_hda_writew(u16 value, u16 *addr) +{ + unsigned int shift_bits = SHIFT_BITS(addr); + writel((readl(ADDR_ALIGN_L(addr)) & ~MASK_REG(addr, 16)) | + ((value) << shift_bits), ADDR_ALIGN_L(addr)); +} + +static u16 tegra_hda_readw(u16 *addr) +{ + return (readl(ADDR_ALIGN_L(addr)) >> SHIFT_BITS(addr)) & MASK(16); +} + +static void tegra_hda_writeb(u8 value, u8 *addr) +{ + writel((readl(ADDR_ALIGN_L(addr)) & ~MASK_REG(addr, 8)) | + ((value) << SHIFT_BITS(addr)), ADDR_ALIGN_L(addr)); +} + +static u8 tegra_hda_readb(u8 *addr) +{ + return (readl(ADDR_ALIGN_L(addr)) >> SHIFT_BITS(addr)) & MASK(8); +} + +static const struct hda_controller_ops tegra_hda_reg_ops = { + .reg_writel = tegra_hda_writel, + .reg_readl = tegra_hda_readl, + .reg_writew = tegra_hda_writew, + .reg_readw = tegra_hda_readw, + .reg_writeb = tegra_hda_writeb, + .reg_readb = tegra_hda_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static int tegra_hda_acquire_irq(struct azx *chip, int do_disconnect) +{ + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + int irq_id = platform_get_irq(tdata->pdev, 0); + + if (devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + dev_err(chip->card->dev, + "unable to grab IRQ %d, disabling device\n", + irq_id); + if (do_disconnect) + snd_card_disconnect(chip->card); + return -1; + } + chip->irq = irq_id; + return 0; +} + +static void tegra_hda_reg_update_bits(void __iomem *base, unsigned int reg, + unsigned int mask, unsigned int val) +{ + unsigned int data; + + data = readl(base + reg); + data &= ~mask; + data |= (val & mask); + writel(data, base + reg); +} + +static void hda_tegra_init(struct hda_tegra_data *tdata) +{ + /*Enable the PCI access */ + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_CONFIG, + NVIDIA_TEGRA_HDA_IPFS_EN_FPCI, + NVIDIA_TEGRA_HDA_IPFS_EN_FPCI); + /* Enable MEM/IO space and bus master */ + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_CMD_OFFSET, 0x507, + NVIDIA_TEGRA_HDA_ENABLE_MEM_SPACE | + NVIDIA_TEGRA_HDA_ENABLE_IO_SPACE | + NVIDIA_TEGRA_HDA_ENABLE_BUS_MASTER | + NVIDIA_TEGRA_HDA_ENABLE_SERR); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_BAR0_INIT_PROGRAM); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_BAR0_FINAL_PROGRAM); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_FPCI_BAR0, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_FPCI_BAR0_START); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_INTR_MASK, + NVIDIA_TEGRA_HDA_IPFS_EN_INTR, + NVIDIA_TEGRA_HDA_IPFS_EN_INTR); + + return; +} + +static void hda_tegra_enable_clocks(struct hda_tegra_data *data) +{ + int i; + + for (i = 0; i < data->platform_clk_count; i++) + clk_prepare_enable(data->platform_clks[i]); +} + +static void hda_tegra_disable_clocks(struct hda_tegra_data *data) +{ + int i; + + for (i = 0; i < data->platform_clk_count; i++) + clk_disable_unprepare(data->platform_clks[i]); +} + +#if CONFIG_PM_SLEEP +/* + * power management + */ +static int tegra_hda_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + if (pm_runtime_active(card->dev)) { + azx_stop_chip(chip); + azx_enter_link_reset(chip); + } + + return 0; +} + +static int tegra_hda_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + + if (pm_runtime_active(card->dev)) { + hda_tegra_init(tdata); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + } + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +#ifdef CONFIG_PM_RUNTIME +static int tegra_hda_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(tdata); + return 0; +} + +static int tegra_hda_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_bus *bus; + struct hda_codec *codec; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + int status; + + hda_tegra_enable_clocks(tdata); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + hda_tegra_init(tdata); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + bus = chip->bus; + if (status && bus) { + list_for_each_entry(codec, &bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, + codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); + + return 0; +} + +#endif /* CONFIG_PM_RUNTIME */ + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(tegra_hda_suspend, tegra_hda_resume) + SET_RUNTIME_PM_OPS(tegra_hda_runtime_suspend, + tegra_hda_runtime_resume, + NULL) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int tegra_hda_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void tegra_hda_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = tegra_hda_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void tegra_hda_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int tegra_hda_free(struct azx *chip) +{ + int i; + + if (chip->running) + pm_runtime_get_noresume(chip->card->dev); + + tegra_hda_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int tegra_hda_dev_free(struct snd_device *device) +{ + return tegra_hda_free(device->device_data); +} + +static const char * const tegra_clk_names[] = { + "hda", + "hda2codec_2x", + "hda2hdmi", +}; + +static int hda_tegra_init_chip(struct azx *chip) +{ + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + struct device *dev = &tdata->pdev->dev; + struct resource *res, *region; + int i; + + tdata->platform_clk_count = ARRAY_SIZE(tegra_clk_names); + tdata->platform_clks = devm_kzalloc(dev, + tdata->platform_clk_count * + sizeof(*tdata->platform_clks), + GFP_KERNEL); + if (!tdata->platform_clks) + return -ENOMEM; + + for (i = 0; i < tdata->platform_clk_count; i++) { + tdata->platform_clks[i] = devm_clk_get(dev, tegra_clk_names[i]); + if (IS_ERR(tdata->platform_clks[i])) + return PTR_ERR(tdata->platform_clks[i]); + } + + res = platform_get_resource(tdata->pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + region = devm_request_mem_region(dev, res->start, + resource_size(res), + tdata->pdev->name); + if (!region) + return -ENOMEM; + + chip->addr = res->start; + chip->remap_addr = devm_ioremap(dev, res->start, resource_size(res)); + if (!chip->remap_addr) + return -ENXIO; + + tdata->remap_config_addr = chip->remap_addr; + chip->remap_addr += NVIDIA_TEGRA_HDA_BAR0_OFFSET; + chip->addr += NVIDIA_TEGRA_HDA_BAR0_OFFSET; + + hda_tegra_enable_clocks(tdata); + + hda_tegra_init(tdata); + + return 0; +} + +static void power_down_all_codecs(struct azx *chip) +{ + /* The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int tegra_hda_first_init(struct azx *chip) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + + err = hda_tegra_init_chip(chip); + if (err) + return err; + + if (tegra_hda_acquire_irq(chip, 0) < 0) + return -EBUSY; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kzalloc(card->dev, + chip->num_streams * sizeof(*chip->azx_dev), + GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "HDA-Tegra"); + strcpy(card->shortname, "HDA-Tegra"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int tegra_hda_create(struct snd_card *card, + int dev, + struct platform_device *pdev, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra_data *tdata) +{ + static struct snd_device_ops ops = { + .dev_free = tegra_hda_dev_free, + }; + struct azx *chip; + int err; + + chip = &tdata->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = dev; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = probe_mask[dev]; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device [card]!\n"); + return err; + } + + return 0; +} + +static unsigned int tegra_driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_PM_RUNTIME; + +static const struct of_device_id tegra_platform_hda_match[] = { + { .compatible = "nvidia,tegra124-hda", .data = &tegra_driver_flags }, + {}, +}; + +static int hda_tegra_probe(struct platform_device *pdev) +{ + static int dev; + struct snd_card *card; + struct azx *chip; + struct hda_tegra_data *tdata; + const struct of_device_id *of_id; + const unsigned int *driver_data; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + + of_id = of_match_device(tegra_platform_hda_match, &pdev->dev); + if (!of_id) + return -EINVAL; + + tdata = devm_kzalloc(&pdev->dev, sizeof(*tdata), GFP_KERNEL); + if (!tdata) + return -ENOMEM; + tdata->pdev = pdev; + chip = &tdata->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + driver_data = of_id->data; + err = tegra_hda_create(card, dev, pdev, *driver_data, + &tegra_hda_reg_ops, tdata); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = tegra_hda_first_init(chip); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, power_save_addr); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + pm_runtime_enable(chip->card->dev); + hda_tegra_disable_clocks(tdata); + power_down_all_codecs(chip); + tegra_hda_notifier_register(chip); + + dev++; + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &azx_pm, + .of_match_table = tegra_platform_hda_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH v2 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 10 Apr '14

10 Apr '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of the module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Kconfig depends on OF and ARCH_TEGRA. Fix warning on 64 bit build. Integrate a bunch of good suggestions from Thierry, including: Add resets property to device tree entry. More consistent/better naming. Remove probe mask that isn't needed. Cleaner/more readable handling of dword write restriction. --- .../bindings/sound/nvidia,tegra30-hda.txt | 27 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 644 +++++++++++++++++++++ 4 files changed, 688 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt new file mode 100644 index 0000000..97f6832 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -0,0 +1,27 @@ +NVIDIA Tegra30 HDA controller + +Required properties: +- compatible : "nvidia,tegra30-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the hda controller. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x0 0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>; /* hda2hdmi */ + <&tegra_car 111>, /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..f17c016 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,21 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "NVIDIA Tegra HD Audio" + depends on OF && ARCH_TEGRA + select SND_HDA + help + Say Y here to support the HDA controller present in NVIDIA + Tegra SoCs + + This options enables support for the HD Audio controller + present in some NVIDIA Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..75f246b --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,644 @@ +/* + * + * Implementation of primary alsa driver code base for NVIDIA Tegra HDA. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/completion.h> +#include <linux/delay.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/of_device.h> +#include <linux/pm_runtime.h> +#include <linux/reboot.h> +#include <linux/slab.h> +#include <linux/time.h> + +#include <sound/core.h> +#include <sound/initval.h> + +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +/* Defines for NVIDIA Tegra HDA support */ +#define HDA_BAR0 0x8000 + +#define HDA_CFG_CMD 0x1004 +#define HDA_CFG_BAR0 0x1010 + +#define HDA_ENABLE_IO_SPACE (1 << 0) +#define HDA_ENABLE_MEM_SPACE (1 << 1) +#define HDA_ENABLE_BUS_MASTER (1 << 2) +#define HDA_ENABLE_SERR (1 << 8) +#define HDA_DISABLE_INTR (1 << 10) +#define HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define HDA_IPFS_CONFIG 0x180 +#define HDA_IPFS_EN_FPCI 0x1 + +#define HDA_IPFS_FPCI_BAR0 0x80 +#define HDA_FPCI_BAR0_START 0x40 + +#define HDA_IPFS_INTR_MASK 0x188 +#define HDA_IPFS_EN_INTR (1 << 16) + +struct hda_tegra { + struct azx chip; + struct device *dev; + struct clk *hda_clk; + struct clk *hda2codec_2x_clk; + struct clk *hda2hdmi_clk; + void __iomem *regs; +}; + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, int type, size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, chip->card->dev, size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +static void hda_tegra_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 hda_tegra_readl(u32 *addr) +{ + return readl(addr); +} + +static void hda_tegra_writew(u16 value, u16 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xffff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u16 hda_tegra_readw(u16 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xffff; +} + +static void hda_tegra_writeb(u8 value, u8 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u8 hda_tegra_readb(u8 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xff; +} + +static const struct hda_controller_ops hda_tegra_ops = { + .reg_writel = hda_tegra_writel, + .reg_readl = hda_tegra_readl, + .reg_writew = hda_tegra_writew, + .reg_readw = hda_tegra_readw, + .reg_writeb = hda_tegra_writeb, + .reg_readb = hda_tegra_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static void hda_tegra_init(struct hda_tegra *hda) +{ + u32 v; + + /*Enable the PCI access */ + v = readl(hda->regs + HDA_IPFS_CONFIG); + v |= HDA_IPFS_EN_FPCI; + writel(v, hda->regs + HDA_IPFS_CONFIG); + + /* Enable MEM/IO space and bus master */ + v = readl(hda->regs + HDA_CFG_CMD); + v &= ~HDA_DISABLE_INTR; + v |= HDA_ENABLE_MEM_SPACE | HDA_ENABLE_IO_SPACE | + HDA_ENABLE_BUS_MASTER | HDA_ENABLE_SERR; + writel(v, hda->regs + HDA_CFG_CMD); + + writel(HDA_BAR0_INIT_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_BAR0_FINAL_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_FPCI_BAR0_START, hda->regs + HDA_IPFS_FPCI_BAR0); + + v = readl(hda->regs + HDA_IPFS_INTR_MASK); + v |= HDA_IPFS_EN_INTR; + writel(v, hda->regs + HDA_IPFS_INTR_MASK); +} + +static int hda_tegra_enable_clocks(struct hda_tegra *data) +{ + int rc; + + rc = clk_prepare_enable(data->hda_clk); + if (rc) + return rc; + rc = clk_prepare_enable(data->hda2codec_2x_clk); + if (rc) + goto disable_hda; + rc = clk_prepare_enable(data->hda2hdmi_clk); + if (rc) + goto disable_codec_2x; + + return 0; + +disable_codec_2x: + clk_disable_unprepare(data->hda2codec_2x_clk); +disable_hda: + clk_disable_unprepare(data->hda_clk); + return rc; +} + +static void hda_tegra_disable_clocks(struct hda_tegra *data) +{ + clk_disable_unprepare(data->hda2hdmi_clk); + clk_disable_unprepare(data->hda2codec_2x_clk); + clk_disable_unprepare(data->hda_clk); +} + +#ifdef CONFIG_PM_SLEEP +/* + * power management + */ +static int hda_tegra_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + if (pm_runtime_active(card->dev)) { + azx_stop_chip(chip); + azx_enter_link_reset(chip); + } + + return 0; +} + +static int hda_tegra_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + if (pm_runtime_active(card->dev)) { + hda_tegra_init(hda); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + } + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +#ifdef CONFIG_PM_RUNTIME +static int hda_tegra_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(hda); + return 0; +} + +static int hda_tegra_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_codec *codec; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + int status; + + hda_tegra_enable_clocks(hda); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + hda_tegra_init(hda); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + if (status && chip->bus) { + list_for_each_entry(codec, &chip->bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, + codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); + + return 0; +} + +#endif /* CONFIG_PM_RUNTIME */ + +static const struct dev_pm_ops hda_tegra_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) + SET_RUNTIME_PM_OPS(hda_tegra_runtime_suspend, + hda_tegra_runtime_resume, + NULL) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int hda_tegra_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void hda_tegra_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = hda_tegra_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void hda_tegra_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int hda_tegra_dev_free(struct snd_device *device) +{ + int i; + struct azx *chip = device->device_data; + + if (chip->running) + pm_runtime_get_noresume(chip->card->dev); + + hda_tegra_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) +{ + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct device *dev = hda->dev; + struct resource *res; + int err; + + hda->hda_clk = devm_clk_get(dev, "hda"); + if (IS_ERR(hda->hda_clk)) + return PTR_ERR(hda->hda_clk); + hda->hda2codec_2x_clk = devm_clk_get(dev, "hda2codec_2x"); + if (IS_ERR(hda->hda2codec_2x_clk)) + return PTR_ERR(hda->hda2codec_2x_clk); + hda->hda2hdmi_clk = devm_clk_get(dev, "hda2hdmi"); + if (IS_ERR(hda->hda2hdmi_clk)) + return PTR_ERR(hda->hda2hdmi_clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + hda->regs = devm_ioremap_resource(dev, res); + if (IS_ERR(chip->remap_addr)) + return PTR_ERR(chip->remap_addr); + + chip->remap_addr = hda->regs + HDA_BAR0; + chip->addr = res->start + HDA_BAR0; + + err = hda_tegra_enable_clocks(hda); + if (err) + return err; + + hda_tegra_init(hda); + + return 0; +} + +/* + * The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ +static void power_down_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + int irq_id = platform_get_irq(pdev, 0); + + err = hda_tegra_init_chip(chip, pdev); + if (err) + return err; + + err = devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip); + if (err) { + dev_err(chip->card->dev, + "unable to request IRQ %d, disabling device\n", + irq_id); + return err; + } + chip->irq = irq_id; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kcalloc(card->dev, chip->num_streams, + sizeof(*chip->azx_dev), GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "tegra-hda"); + strcpy(card->shortname, "tegra-hda"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int hda_tegra_create(struct snd_card *card, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra *hda) +{ + static struct snd_device_ops ops = { + .dev_free = hda_tegra_dev_free, + }; + struct azx *chip; + int err; + + chip = &hda->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = 0; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = -1; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device\n"); + return err; + } + + return 0; +} + +static const struct of_device_id hda_tegra_match[] = { + { .compatible = "nvidia,tegra30-hda" }, + {}, +}; + +static int hda_tegra_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct azx *chip; + struct hda_tegra *hda; + const struct of_device_id *of_id; + int err; + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_PM_RUNTIME; + + of_id = of_match_device(hda_tegra_match, &pdev->dev); + if (!of_id) + return -ENODEV; + + hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); + if (!hda) + return -ENOMEM; + hda->dev = &pdev->dev; + chip = &hda->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + err = hda_tegra_create(card, driver_flags, &hda_tegra_ops, hda); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = hda_tegra_first_init(chip, pdev); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, &power_save); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + pm_runtime_enable(chip->card->dev); + hda_tegra_disable_clocks(hda); + power_down_all_codecs(chip); + hda_tegra_notifier_register(chip); + + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = "tegra-hda", + .pm = &hda_tegra_pm, + .of_match_table = hda_tegra_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH v2 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 10 Apr '14

10 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- No changes since v1 --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3377,6 +3378,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH] ASoC: rt5640: add more settings for rt5639
by Oder Chiou 10 Apr '14

10 Apr '14

The patch adds the the list of OF compatible strings, the binding document and the list of I2C device IDs for rt5639. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5639.txt | 48 ++++++++++ sound/soc/codecs/rt5640.c | 103 ++++++++++++++++++++- 2 files changed, 147 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5639.txt diff --git a/Documentation/devicetree/bindings/sound/rt5639.txt b/Documentation/devicetree/bindings/sound/rt5639.txt new file mode 100644 index 0000000..d63c662 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5639.txt @@ -0,0 +1,48 @@ +RT5639 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5639". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential + Boolean. Indicate MIC1/2 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes): + + * DMIC1 + * DMIC2 + * MICBIAS1 + * IN1P + * IN1R + * IN2P + * IN2R + * HPOL + * HPOR + * LOUTL + * LOUTR + * SPOLP + * SPOLN + * SPORP + * SPORN + +Example: + +rt5639 { + compatible = "realtek,rt5639"; + reg = <0x1c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index d672f44..d7d6cad 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1,5 +1,5 @@ /* - * rt5640.c -- RT5640 ALSA SoC audio codec driver + * rt5640.c -- RT5640/RT5639 ALSA SoC audio codec driver * * Copyright 2011 Realtek Semiconductor Corp. * Author: Johnny Hsu <johnnyhsu(a)realtek.com> @@ -2131,6 +2131,47 @@ static struct snd_soc_dai_driver rt5640_dai[] = { }, }; +static struct snd_soc_dai_driver rt5639_dai[] = { + { + .name = "rt5639-aif1", + .id = RT5640_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, + { + .name = "rt5639-aif2", + .id = RT5640_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, +}; + static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .probe = rt5640_probe, .remove = rt5640_remove, @@ -2168,12 +2209,38 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static const struct i2c_device_id rt5639_i2c_id[] = { + { "rt5639", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5640", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5640_of_match); + +static const struct of_device_id rt5639_of_match[] = { + { .compatible = "realtek,rt5639", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5639_of_match); +#endif + #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); + +static struct acpi_device_id rt5639_acpi_match[] = { + { "INT33CA", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5639_acpi_match); #endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) @@ -2293,8 +2360,18 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); + regmap_read(rt5640->regmap, RT5640_RESET, &val); + switch (val) { + case RT5640_RESET_ID: + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); + break; + case RT5639_RESET_ID: + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5639_dai, ARRAY_SIZE(rt5639_dai)); + break; + } + if (ret < 0) goto err; @@ -2315,6 +2392,9 @@ static struct i2c_driver rt5640_i2c_driver = { .name = "rt5640", .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(rt5640_of_match), +#endif }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, @@ -2322,6 +2402,21 @@ static struct i2c_driver rt5640_i2c_driver = { }; module_i2c_driver(rt5640_i2c_driver); -MODULE_DESCRIPTION("ASoC RT5640 driver"); +static struct i2c_driver rt5639_i2c_driver = { + .driver = { + .name = "rt5639", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5639_acpi_match), +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(rt5639_of_match), +#endif + }, + .probe = rt5640_i2c_probe, + .remove = rt5640_i2c_remove, + .id_table = rt5639_i2c_id, +}; +module_i2c_driver(rt5639_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5640/RT5639 driver"); MODULE_AUTHOR("Johnny Hsu <johnnyhsu(a)realtek.com>"); MODULE_LICENSE("GPL v2"); -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] move alc892 to patch_alc882
by Kailang 10 Apr '14

10 Apr '14

Hi Takashi, The original was patch to patch_alc662. Move it into patch_alc882. Thanks. BR, Kailang

3 21

[alsa-devel] [PATCH linux-next] ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
by Tim Gardner 09 Apr '14

09 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305133 Malfunctioning or slow devices can cause a flood of dmesg SPAM. I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour of prior art in sound/usb/pcm.c. WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit + if (printk_ratelimit() && Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Eldad Zack <eldad(a)fogrefinery.com> Cc: Daniel Mack <zonque(a)gmail.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49de5c1..131336d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1501,7 +1501,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, * The error should be lower than 2ms since the estimate relies * on two reads of a counter updated every ms. */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + if (printk_ratelimit() && + abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) dev_dbg(&subs->dev->dev, "delay: estimated %d, actual %d\n", est_delay, subs->last_delay); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Add support for multi register mux
by Arun Shamanna Lakshmi 09 Apr '14

09 Apr '14

1. Modify soc_enum struct to handle pointers for reg and mask 2. Add dapm get and put APIs for multi register one hot encoded mux 3. Update snd_soc_dapm_update struct to support multiple reg update Signed-off-by: Arun S L <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 20 ++++- include/sound/soc.h | 33 +++++++-- sound/soc/soc-core.c | 12 +-- sound/soc/soc-dapm.c | 181 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 203 insertions(+), 43 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..c255349 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_ONEHOT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_onehot, \ + .put = snd_soc_dapm_put_enum_onehot, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -352,6 +358,9 @@ struct device; /* regulator widget flags */ #define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* maximum registers to update */ +#define SND_SOC_DAPM_UPDATE_MAX_REG 3 + struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; @@ -378,6 +387,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +603,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[SND_SOC_DAPM_UPDATE_MAX_REG]; + int mask[SND_SOC_DAPM_UPDATE_MAX_REG]; + int val[SND_SOC_DAPM_UPDATE_MAX_REG]; + unsigned int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..dcdf1cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,24 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1, .type = SND_SOC_ENUM_BINARY } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1, .type = SND_SOC_ENUM_BINARY } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, \ + .type = SND_SOC_ENUM_ONEHOT } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +299,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -326,6 +335,16 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +/* + * enum snd_soc_enum_type - Type of ASoC enum control + * @SND_SOC_ENUM_BINARY: soc_enum type for SINGLE, DOUBLE or VIRTUAL mux + * @SND_SOC_ENUM_ONEHOT: soc_enum type for ONEHOT encoding mux + */ +enum snd_soc_enum_type { + SND_SOC_ENUM_BINARY = 0, + SND_SOC_ENUM_ONEHOT = 1, +}; + struct device_node; struct snd_jack; struct snd_soc_card; @@ -1098,13 +1117,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + enum snd_soc_enum_type type; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..cf29722 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..be9cd63 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -511,13 +511,26 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, const struct snd_kcontrol_new *kcontrol) { struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; + unsigned int val, item, bit_pos = 0; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); + if (e->reg[0] != SND_SOC_NOPM) { + if (e->type == SND_SOC_ENUM_ONEHOT) { + for (i = 0; i < e->num_regs; i++) { + soc_widget_read(dest, e->reg[i], &val); + val = val & e->mask[i]; + if (val != 0) { + bit_pos = ffs(val) + + (8 * dest->codec->val_bytes * i); + break; + } + } + item = snd_soc_enum_val_to_item(e, bit_pos); + } else { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; + item = snd_soc_enum_val_to_item(e, val); + } } else { /* since a virtual mux has no backing registers to * decide which path to connect, it will try to match @@ -1553,8 +1566,8 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; - int ret; + unsigned int wi, i; + int ret = 0; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; @@ -1575,11 +1588,16 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); - if (ret < 0) - dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", - w->name, ret); + /* dapm update for multiple registers */ + for (i = 0; i < update->num_regs; i++) { + ret = soc_widget_update_bits_locked(w, update->reg[i], + update->mask[i], update->val[i]); + if (ret < 0) { + dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", + w->name, ret); + break; + } + } for (wi = 0; wi < wlist->num_widgets; wi++) { w = wlist->widgets[wi]; @@ -2866,10 +2884,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; - + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2921,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2963,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +3003,112 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_onehot - dapm enumerated onehot mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = 0, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = ffs(val) + (8 * codec->val_bytes * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_onehot); + +/** + * snd_soc_dapm_put_enum_onehot - dapm enumerated onehot mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + unsigned int i, reg_val = 0, update_idx = 0; + struct snd_soc_dapm_update update; + int ret = 0; + + if (item[0] >= e->items || e->num_regs > SND_SOC_DAPM_UPDATE_MAX_REG) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value) { + /* get the register index and value to set */ + reg_idx = (value - 1) / (8 * codec->val_bytes); + bit_pos = (value - 1) % (8 * codec->val_bytes); + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + /* set the selected register */ + update.reg[e->num_regs - 1] = e->reg[reg_idx]; + update.mask[e->num_regs - 1] = e->mask[reg_idx]; + update.val[e->num_regs - 1] = reg_val; + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + update.reg[update_idx] = e->reg[i]; + update.mask[update_idx] = e->mask[i]; + update.val[update_idx++] = 0; + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.num_regs = e->num_regs; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_onehot); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 3

[alsa-devel] ASoC updates for v3.15
by Mark Brown 09 Apr '14

09 Apr '14

The following changes since commit 455c6fdbd219161bd09b1165f11699d6d73de11c: Linux 3.14 (2014-03-30 20:40:15 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-5 for you to fetch changes up to ab5d6fbdb71c183add7431243c7f615c42b643bb: Merge remote-tracking branches 'asoc/fix/rt5640', 'asoc/fix/samsung', 'asoc/fix/tlv320aic23' and 'asoc/fix/warn' into asoc-linus (2014-04-08 21:22:14 +0100) Some extra merges coming in due to the way my script ensures that previously sent fixes are included in my branch still not interacting well with signed tags, I'll need to figure that out. ---------------------------------------------------------------- ASoC: Fixes for v3.15 A smattering of device specific fixes, nothing stands out here except for the multiplatform fixes for Samsung and the device IDs being added by Stephen Warren - there's no real code changes from those and they give better robustness to the enumeration with DT. ---------------------------------------------------------------- Axel Lin (3): ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value ASoC: cs42l52: Fix mask for REVID ASoC: da732x: Print correct major id Christoph Jaeger (1): ASoC: alc56(23|32): fix undefined return value of probing code Mark Brown (6): Merge tag 'asoc-v3.15' into asoc-linus Merge tag 'asoc-v3.15-2' into asoc-linus Merge tag 'asoc-v3.15-3' into asoc-linus Merge tag 'asoc-v3.15-4' into asoc-linus Merge remote-tracking branches 'asoc/fix/alc5632', 'asoc/fix/cs42l52', 'asoc/fix/cs42xxx8', 'asoc/fix/da732x', 'asoc/fix/davinci', 'asoc/fix/fsl-sai', 'asoc/fix/fsl-ssi' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5640', 'asoc/fix/samsung', 'asoc/fix/tlv320aic23' and 'asoc/fix/warn' into asoc-linus Markus Pargmann (1): ASoC: fsl-ssi: Update binding documentation Nicolin Chen (2): ASoC: fsl_sai: Add isr to deal with error flag ASoC: fsl_sai: Fix Bit Clock Polarity configurations Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix bit clock polarity settings Sachin Kamat (1): ASoC: samsung: Fix build on multiplatform Stephen Warren (5): ASoC: rt5640: add an of_match table ASoC: alc5632: fix uninit var in alc5632_probe() ASoC: alc5632: add an of_match table ASoC: max98090: add an of_match table ASoC: tlv320aic23: add an of_match table .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 8 + .../bindings/sound/armada-370db-audio.txt | 27 + .../devicetree/bindings/sound/cs42xx8.txt | 28 + Documentation/devicetree/bindings/sound/da9055.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- .../devicetree/bindings/sound/mvebu-audio.txt | 1 + .../devicetree/bindings/sound/pcm512x.txt | 30 + .../devicetree/bindings/sound/renesas,rsnd.txt | 105 ++ .../devicetree/bindings/sound/simple-card.txt | 65 +- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 + .../devicetree/bindings/sound/tdm-slot.txt | 20 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 30 + .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - .../devicetree/bindings/sound/widgets.txt | 20 + MAINTAINERS | 7 + arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +- arch/arm/mach-shmobile/board-bockw.c | 16 +- arch/arm/mach-shmobile/board-kzm9g.c | 4 +- arch/arm/mach-shmobile/board-mackerel.c | 6 +- arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- drivers/base/regmap/regmap.c | 12 + drivers/misc/atmel-ssc.c | 6 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + include/linux/atmel-ssc.h | 1 + include/linux/mfd/arizona/registers.h | 6 +- include/linux/platform_data/adau1977.h | 45 + include/linux/platform_data/asoc-s3c.h | 3 +- include/linux/platform_data/asoc-s3c24xx_simtec.h | 3 +- include/linux/platform_data/davinci_asp.h | 4 +- include/linux/regmap.h | 9 + include/sound/rcar_snd.h | 36 +- include/sound/simple_card.h | 6 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 40 +- include/sound/soc.h | 152 +- include/trace/events/hswadsp.h | 384 +++++ include/trace/events/intel-sst.h | 148 ++ sound/core/pcm_lib.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 119 +- sound/soc/codecs/Kconfig | 195 ++- sound/soc/codecs/Makefile | 39 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 154 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/adau1373.c | 39 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 159 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 24 +- sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 250 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 123 +- sound/soc/codecs/alc5632.c | 58 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 99 +- sound/soc/codecs/cs42l52.c | 111 +- sound/soc/codecs/cs42l52.h | 2 +- sound/soc/codecs/cs42l73.c | 72 +- sound/soc/codecs/cs42xx8-i2c.c | 64 + sound/soc/codecs/cs42xx8.c | 601 +++++++ sound/soc/codecs/cs42xx8.h | 238 +++ sound/soc/codecs/da7210.c | 28 +- sound/soc/codecs/da7213.c | 159 +- sound/soc/codecs/da732x.c | 199 +-- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 100 +- sound/soc/codecs/isabelle.c | 19 +- sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 47 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 47 +- sound/soc/codecs/max98090.c | 188 ++- sound/soc/codecs/max98090.h | 1 - sound/soc/codecs/max98095.c | 60 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 30 +- sound/soc/codecs/ml26124.c | 22 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 84 +- sound/soc/codecs/rt5640.c | 95 +- sound/soc/codecs/sgtl5000.c | 18 +- sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sirf-audio-codec.c | 524 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 46 +- sound/soc/codecs/ssm2518.c | 24 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 180 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta32x.c | 14 - sound/soc/codecs/sta529.c | 15 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 66 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 79 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 7 +- sound/soc/codecs/tlv320aic31xx.c | 1280 ++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 +++ sound/soc/codecs/tlv320aic32x4.c | 234 ++- sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 35 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 43 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 25 +- sound/soc/codecs/wm5100.c | 43 +- sound/soc/codecs/wm5102.c | 32 +- sound/soc/codecs/wm5110.c | 22 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 11 +- sound/soc/codecs/wm8580.c | 9 +- sound/soc/codecs/wm8711.c | 8 +- sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8737.c | 56 +- sound/soc/codecs/wm8741.c | 40 +- sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 12 +- sound/soc/codecs/wm8770.c | 6 - sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 10 +- sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 118 +- sound/soc/codecs/wm8904.c | 86 +- sound/soc/codecs/wm8940.c | 26 +- sound/soc/codecs/wm8955.c | 19 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 23 +- sound/soc/codecs/wm8962.c | 87 +- sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 10 +- sound/soc/codecs/wm8978.c | 38 +- sound/soc/codecs/wm8983.c | 51 +- sound/soc/codecs/wm8985.c | 46 +- sound/soc/codecs/wm8988.c | 70 +- sound/soc/codecs/wm8990.c | 49 +- sound/soc/codecs/wm8991.c | 52 +- sound/soc/codecs/wm8993.c | 73 +- sound/soc/codecs/wm8994.c | 55 +- sound/soc/codecs/wm8995.c | 50 +- sound/soc/codecs/wm8996.c | 87 +- sound/soc/codecs/wm8997.c | 29 +- sound/soc/codecs/wm9081.c | 34 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 80 +- sound/soc/davinci/davinci-mcasp.c | 219 ++- sound/soc/davinci/edma-pcm.c | 57 + sound/soc/davinci/edma-pcm.h | 25 + sound/soc/fsl/Kconfig | 14 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 34 +- sound/soc/fsl/fsl_sai.c | 397 +++-- sound/soc/fsl/fsl_sai.h | 63 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/fsl_utils.c | 27 + sound/soc/fsl/fsl_utils.h | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/imx-ssi.c | 2 + sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 385 +++-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 108 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 22 +- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 60 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 67 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/ac97.c | 6 +- sound/soc/samsung/dma.h | 6 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/pcm.c | 5 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 3 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 229 ++- sound/soc/sh/rcar/core.c | 426 +++-- sound/soc/sh/rcar/gen.c | 113 +- sound/soc/sh/rcar/rsnd.h | 206 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 727 ++++++++ sound/soc/sh/rcar/ssi.c | 388 ++--- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 603 ++++--- sound/soc/soc-dapm.c | 465 ++--- sound/soc/soc-io.c | 99 +- sound/soc/soc-jack.c | 5 +- sound/soc/soc-pcm.c | 112 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 17 +- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_wm9712.c | 17 +- 292 files changed, 21319 insertions(+), 5575 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) create mode 100644 sound/soc/kirkwood/armada-370-db.c delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c

2 1

[alsa-devel] [PATCH] ALSA: hda - Make full_reset boolean
by Thierry Reding 09 Apr '14

09 Apr '14

From: Thierry Reding <treding(a)nvidia.com> The full_reset argument to azx_init_chip() carries boolean rather than numerical information, so update the type to reflect that. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- Changes in v2: - update prototype in hda_controller.h sound/pci/hda/hda_controller.c | 8 ++++---- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 97993e17f46a..98d31d153703 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1604,7 +1604,7 @@ static void azx_exit_link_reset(struct azx *chip) } /* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +static int azx_reset(struct azx *chip, bool full_reset) { if (!full_reset) goto __skip; @@ -1701,7 +1701,7 @@ static void azx_int_clear(struct azx *chip) /* * reset and start the controller registers */ -void azx_init_chip(struct azx *chip, int full_reset) +void azx_init_chip(struct azx *chip, bool full_reset) { if (chip->initialized) return; @@ -1841,7 +1841,7 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); #ifdef CONFIG_PM if (chip->initialized) { struct azx_pcm *p; @@ -1948,7 +1948,7 @@ int azx_codec_create(struct azx *chip, const char *model, * get back to the sanity state. */ azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); } } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 1d2e3be2bae6..baf0e77330af 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ int azx_alloc_stream_pages(struct azx *chip); void azx_free_stream_pages(struct azx *chip); /* Low level azx interface */ -void azx_init_chip(struct azx *chip, int full_reset); +void azx_init_chip(struct azx *chip, bool full_reset); void azx_stop_chip(struct azx *chip); void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77ca894f8284..d6bca62ef387 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -636,7 +636,7 @@ static int azx_resume(struct device *dev) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -689,7 +689,7 @@ static int azx_runtime_resume(struct device *dev) status = azx_readw(chip, STATESTS); azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); bus = chip->bus; if (status && bus) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Make full_reset boolean
by Thierry Reding 09 Apr '14

09 Apr '14

From: Thierry Reding <treding(a)nvidia.com> The full_reset argument to azx_init_chip() carries boolean rather than numerical information, so update the type to reflect that. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_controller.c | 8 ++++---- sound/pci/hda/hda_intel.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 97993e17f46a..98d31d153703 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1604,7 +1604,7 @@ static void azx_exit_link_reset(struct azx *chip) } /* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +static int azx_reset(struct azx *chip, bool full_reset) { if (!full_reset) goto __skip; @@ -1701,7 +1701,7 @@ static void azx_int_clear(struct azx *chip) /* * reset and start the controller registers */ -void azx_init_chip(struct azx *chip, int full_reset) +void azx_init_chip(struct azx *chip, bool full_reset) { if (chip->initialized) return; @@ -1841,7 +1841,7 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); #ifdef CONFIG_PM if (chip->initialized) { struct azx_pcm *p; @@ -1948,7 +1948,7 @@ int azx_codec_create(struct azx *chip, const char *model, * get back to the sanity state. */ azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); } } } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77ca894f8284..d6bca62ef387 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -636,7 +636,7 @@ static int azx_resume(struct device *dev) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -689,7 +689,7 @@ static int azx_runtime_resume(struct device *dev) status = azx_readw(chip, STATESTS); azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); bus = chip->bus; if (status && bus) { -- 1.9.1

2 2

[alsa-devel] [PATCH v3 0/2] ASoC: fsl_sai: Add bus clock and mclks with clock controls
by Nicolin Chen 09 Apr '14

09 Apr '14

This series of patches add clock controls to fsl_sai driver and updates the vf610.dtsi accordingly. Changelog v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. Nicolin Chen (2): ASoC: fsl_sai: Add clock controls for SAI ARM: dts: Append clock bindings for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 8 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 51 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 60 insertions(+), 9 deletions(-) -- 1.8.4

2 6

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 09 Apr '14

09 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0283, SID: 0x10280667), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0d72dbf..14ae979 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4618,6 +4618,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] sound: dmasound: use module_platform_driver_probe()
by Christoph Jaeger 09 Apr '14

09 Apr '14

Eliminate boilerplate code by using module_platform_driver_probe(). Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/oss/dmasound/dmasound_paula.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 87910e9..c2d45a5 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -733,19 +733,7 @@ static struct platform_driver amiga_audio_driver = { }, }; -static int __init amiga_audio_init(void) -{ - return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe); -} - -module_init(amiga_audio_init); - -static void __exit amiga_audio_exit(void) -{ - platform_driver_unregister(&amiga_audio_driver); -} - -module_exit(amiga_audio_exit); +module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:amiga-audio"); -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: au1x00: use module_platform_driver()
by Christoph Jaeger 09 Apr '14

09 Apr '14

Eliminate boilerplate code by using module_platform_driver(). Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/mips/au1x00.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index a7cc49e..d10ef76 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -725,15 +725,4 @@ struct platform_driver au1000_ac97c_driver = { .remove = au1000_ac97_remove, }; -static int __init au1000_ac97_load(void) -{ - return platform_driver_register(&au1000_ac97c_driver); -} - -static void __exit au1000_ac97_unload(void) -{ - platform_driver_unregister(&au1000_ac97c_driver); -} - -module_init(au1000_ac97_load); -module_exit(au1000_ac97_unload); +module_platform_driver(au1000_ac97c_driver); -- 1.9.0

2 1

[alsa-devel] 2.1 audio not working in Kubuntu 13.10
by Tomas Black 09 Apr '14

09 Apr '14

Dear friends, I have a Dell L702X laptop which comes with a 2.1 sound configuration - 2 speakers + subwoofer from JBL. Sound is suberb in Windows, but I can't get the subwoofer to work in Linux! Here's the information I think you need: &&&&&@dell-xps:~$ aplay -l **** Lista de Dispositivos PLAYBACK Hardware **** placa 0: PCH [HDA Intel PCH], dispositivo 0: ALC665 Analog [ALC665 Analog] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 placa 0: PCH [HDA Intel PCH], dispositivo 1: ALC665 Digital [ALC665 Digital] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 placa 0: PCH [HDA Intel PCH], dispositivo 3: HDMI 0 [HDMI 0] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 #######@dell-xps:~$ aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC665 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct sample mixing device dmix:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions ))))))))@dell-xps:~$ lspci -v | grep -A7 -i audio 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) Subsystem: Dell Device 04b8 Flags: bus master, fast devsel, latency 0, IRQ 54 Memory at f3c00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b5) (prog-if 00 [Normal decode]) ^^^^^^@dell-xps:~$ cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC665 Codec: Intel CougarPoint HDMI I've installed Realtek audio drivers and pretty much tried all tweaks on alsa .conf files a pulse .conf files. As I said, currently the stereo sound is good on both speakers, but how can I get the subwoofer to work? THanks a million in advance, Tomas

2 3

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 286 ++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 3 + 2 files changed, 213 insertions(+), 76 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..d672f44 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -399,18 +399,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), @@ -442,6 +437,15 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +559,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +695,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +750,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -1103,26 +1152,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1134,21 +1172,14 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1198,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1208,10 +1231,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1435,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1485,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1497,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1521,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1901,6 +2013,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 08 Apr '14

08 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3377,6 +3378,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH Resend 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 08 Apr '14

08 Apr '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ice1712: Fix boundary checks in PCM pointer ops
by Takashi Iwai 08 Apr '14

08 Apr '14

PCM pointer callbacks in ice1712 driver check the buffer size boundary wrongly between bytes and frames. This leads to PCM core warnings like: snd_pcm_update_hw_ptr0: 105 callbacks suppressed ALSA pcm_lib.c:352 BUG: pcmC3D0c:0, pos = 5461, buffer size = 5461, period size = 2730 This patch fixes these checks to be placed after the proper unit conversions. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/ice1712.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index da005493f060..d9b9e4595f17 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -685,9 +685,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pointer(struct snd_pcm_substream * if (!(snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL) & 1)) return 0; ptr = runtime->buffer_size - inw(ice->ddma_port + 4); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substream *substream) @@ -704,9 +705,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substrea addr = ICE1712_DSC_ADDR0; ptr = snd_ice1712_ds_read(ice, substream->number * 2, addr) - ice->playback_con_virt_addr[substream->number]; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *substream) @@ -717,9 +719,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *s if (!(snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL) & 1)) return 0; ptr = inl(ICEREG(ice, CONCAP_ADDR)) - ice->capture_con_virt_addr; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback = { @@ -1116,9 +1119,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pro_pointer(struct snd_pcm_substre if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_PLAYBACK_START)) return 0; ptr = ice->playback_pro_size - (inw(ICEMT(ice, PLAYBACK_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substream *substream) @@ -1129,9 +1133,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substrea if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_CAPTURE_START_SHADOW)) return 0; ptr = ice->capture_pro_size - (inw(ICEMT(ice, CAPTURE_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback_pro = { -- 1.9.1

1 0

[alsa-devel] (JOB) ALSA, ASoC driver development
by Adam Ćwiek 08 Apr '14

08 Apr '14

Hi, I am developing multi channel audio dsp platform for high level audio processing. Now I am searching for someone, who can deal with audio ALSA, ASoC driver development (of course paid offer). I am electronic engeneer, so I will take care of hardware. I am going to use audio codecs, like wolfsons WM8510, CS42448 from Cirrus Logic or PCM3168 from Texas. My goal is to be able to send / receive as many analog inputs/outputs as possible. As mainboard i want to use ready made modules as OLinuXino , Raspberry Pi, or BeagleBone. Looking forward to hearing from you. Adam Cwiek

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix Bit Clock Polarity configurations
by Nicolin Chen 08 Apr '14

08 Apr '14

The BCP bit in TCR4/RCR4 register rules as followings: 0 Bit clock is active high with drive outputs on rising edge and sample inputs on falling edge. 1 Bit clock is active low with drive outputs on falling edge and sample inputs on rising edge. For all formats currently supported in the fsl_sai driver, they're exactly sending data on the falling edge and sampling on the rising edge. However, the driver clears this BCP bit for all of them which results click noise when working with SGTL5000 and big noise with WM8962. Thus this patch corrects the BCP settings for all the formats here to fix the nosie issue. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 99051c7..9bbebea 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -180,7 +180,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_LEFT_J: @@ -188,7 +188,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one word length for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: @@ -198,7 +198,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; @@ -208,7 +208,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one bit for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 365 +++++++++++++++++++++++++++++++--------------- sound/soc/codecs/rt5640.h | 3 + 2 files changed, 250 insertions(+), 118 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..f5b9ecc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -361,24 +361,23 @@ static unsigned int bst_tlv[] = { static const char * const rt5640_data_select[] = { "Normal", "left copy to right", "right copy to left", "Swap"}; -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); /* Class D speaker gain ratio */ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, +static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); static const struct snd_kcontrol_new rt5640_snd_controls[] = { @@ -387,11 +386,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* Headphone Output Volume */ SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ SOC_DOUBLE("OUT Playback Switch", RT5640_OUTPUT, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), @@ -399,49 +400,54 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, - RT5640_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, - RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, + bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, + bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, - RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, - 31, 1, in_vol_tlv), + RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ SOC_DOUBLE("ADC Capture Switch", RT5640_ADC_DIG_VOL, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC Capture Volume", RT5640_ADC_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5640_ADC_DATA, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("ADC Boost Gain", RT5640_ADC_BST_VOL, - RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, - 3, 0, adc_bst_tlv), + RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + /* Class D speaker gain ratio */ SOC_ENUM("Class D SPK Ratio Control", rt5640_clsd_spk_ratio_enum), + /* Data Swap */ SOC_ENUM("ADC IF1 Data Switch", rt5640_if1_adc_enum), SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +561,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +697,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +752,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -753,8 +804,7 @@ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = @@ -764,8 +814,7 @@ static const char * const rt5640_stereo_adc2_src[] = { "DMIC1", "DMIC2", "DIG MIX" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = @@ -776,8 +825,7 @@ static const char * const rt5640_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADCL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = @@ -787,8 +835,7 @@ static const char * const rt5640_mono_adc_l2_src[] = { "DMIC L1", "DMIC L2", "Mono DAC MIXL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = @@ -798,8 +845,7 @@ static const char * const rt5640_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADCR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = @@ -809,8 +855,7 @@ static const char * const rt5640_mono_adc_r2_src[] = { "DMIC R1", "DMIC R2", "Mono DAC MIXR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = @@ -1103,26 +1148,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1132,23 +1166,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), /* DACs */ - SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), - SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, + 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1193,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1199,6 +1217,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, &hp_r_enable_control), SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), + /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1208,10 +1227,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1431,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1481,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1493,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1517,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1551,8 +1659,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; @@ -1901,6 +2008,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: core: Allow snd_soc_update_bits use 32 bits register
by Bard Liao 08 Apr '14

08 Apr '14

Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 6 +++--- sound/soc/soc-core.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..b18cf27 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,12 +469,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..7e2734d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; @@ -2357,7 +2357,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2382,7 +2382,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); * * Returns 1 for change else 0. */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2897,7 +2897,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; int err = 0; - unsigned short val, val_mask, val2 = 0; + unsigned int val, val_mask, val2 = 0; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] + min) & mask; -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by mick 08 Apr '14

08 Apr '14

Clemens, please help me out: are you asking me personally to test or is this a general call for the group? If you need me to test, what version of kernel sources do I patch? I am probably not running the latest. Forgive the questions, I do not routinely do these things. Also: Your patch looks quite different from the solution I pointed out. Why? Did you test it yourself? In particular the assertion: > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. is quite different from the assertion "the 3rd byte of the feadback value is getting corrupted and probably changing when it should not " from http://forums.slimdevices.com/showthread.php?93310-Async-USB-Dac-Experiment…. Also, you made it quite explicit for the TEAC UD-H01 but I think it should be more general. I have the Teac A-H01 and it also has the problem. As far as I gather from the various discussions on the Web it probably is a general Tenor 8802 issue. For instance with Nuforce DAC-9 here: http://www.spinics.net/linux/fedora/alsa-user/msg11387.html Mick On Mon, 07 Apr 2014 22:09:21 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: > Please test this patch: > > --8<----------------------------------------------------------------->8-- > ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data > > The TEAC UD-H01 firmware sends wrong feedback frequency values, thus > causing the PC to send the samples at a wrong rate, which results in > clicks and crackles in the output. > > Add a workaround to detect and fix the corruption. > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Tested-by: Mick <mick37(a)gmx.de> > Cc: <stable(a)vger.kernel.org> > --- > sound/usb/card.h | 1 + > sound/usb/endpoint.c | 15 ++++++++++++++- > 2 files changed, 15 insertions(+), 1 deletion(-) > > diff --git a/sound/usb/card.h b/sound/usb/card.h > index 9867ab8..97acb90 100644 > --- a/sound/usb/card.h > +++ b/sound/usb/card.h > @@ -92,6 +92,7 @@ struct snd_usb_endpoint { > unsigned int curframesize; /* current packet size in frames (for > capture) */ > unsigned int syncmaxsize; /* sync endpoint packet size */ > unsigned int fill_max:1; /* fill max packet size always */ > + unsigned int udh01_fb_quirk:1; /* corrupted feedback data */ > unsigned int datainterval; /* log_2 of data packet interval */ > unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ > unsigned char silence_value; > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index e70a87e..7b400eb 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -471,6 +471,10 @@ struct snd_usb_endpoint > *snd_usb_add_endpoint(struct snd_usb_audio *chip, > ep->syncinterval = 3; > > ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); > + > + if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && > + ep->syncmaxsize == 4) > + ep->udh01_fb_quirk = 1; > } > > list_add_tail(&ep->list, &chip->ep_list); > @@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct > snd_usb_endpoint *ep, > if (f == 0) > return; > > - if (unlikely(ep->freqshift == INT_MIN)) { > + if (unlikely(ep->udh01_fb_quirk)) { > + /* > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. > + */ > + if (f < ep->freqn - 0x8000) > + f += 0x10000; > + else if (f > ep->freqn + 0x8000) > + f -= 0x10000; > + } else if (unlikely(ep->freqshift == INT_MIN)) { > /* > * The first time we see a feedback value, determine its format > * by shifting it left or right until it matches the nominal -- Erstellt mit Operas E-Mail-Modul: http://www.opera.com/mail/

2 1

Re: [alsa-devel] [PATCH] ALSA: hda - Do not assign streams in reverse order
by Takashi Iwai 08 Apr '14

08 Apr '14

At Mon, 7 Apr 2014 22:36:38 +0300, Anssi Hannula wrote: > > Currently stream numbers are assigned in reverse order. > > Unfortunately commit 7546abfb8e1f9933b5 ("ALSA: hda - Increment > default stream numbers for AMD HDMI controllers") assumed this was not > the case (specifically, it had the "old cards had single device only" > => "extra unused stream numbers do not matter" assumption), causing > non-working audio regressions for AMD Radeon HDMI users on v3.14. > > Change the stream numbers to be assigned in increasing order instead. > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=77002 > Reported-by: Christian Güdel <cg(a)dmesg.ch> > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Tested-by: Christian Güdel <cg(a)dmesg.ch> > Cc: Alex Deucher <alexander.deucher(a)amd.com> > Cc: <stable(a)vger.kernel.org> # 3.14 > --- > > Takashi, please check I didn't miss anything (I'm not that familiar > with this low-level HDA code). :) In that case, let's revert the commit instead. There was a reason to behave in that way, although I don't remember well. That said, this change might lead to another regression. Thus, if a fix is needed for stable, the revert would be the best choice as a first aid. Of course, this change is good to have, so let's implement this for 3.15 and see whether it works well for all nasty controllers. Besides, this patch can't be applied any longer to the current code base since it's been rewritten a lot; most of codes have been moved to another file hda_controller.c. Could you rebase the patch for the latest tree? Meanwhile I'll prepare a revert that is applied only to 3.14.x. thanks, Takashi > > > sound/pci/hda/hda_intel.c | 9 +++++---- > 1 file changed, 5 insertions(+), 4 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index e354ab1ec20f..7a7b7b10dddf 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1856,13 +1856,14 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) > struct azx_dev *azx_dev = &chip->azx_dev[dev]; > dsp_lock(azx_dev); > if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { > - res = azx_dev; > - if (res->assigned_key == key) { > - res->opened = 1; > - res->assigned_key = key; > + if (azx_dev->assigned_key == key) { > + azx_dev->opened = 1; > + azx_dev->assigned_key = key; > dsp_unlock(azx_dev); > return azx_dev; > } > + if (!res) > + res = azx_dev; > } > dsp_unlock(azx_dev); > } > -- > 1.8.4.5 >

2 4

Re: [alsa-devel] Audio codec with multiple I2C devices
by Lars-Peter Clausen 08 Apr '14

08 Apr '14

On 04/07/2014 09:29 PM, Gilles wrote: > On Apr 7, 2014, at 12:03 , Lars-Peter Clausen <lars(a)metafoo.de> wrote: > >> On 04/07/2014 08:32 PM, Gilles wrote: >> [...] >>> Doesn't it make perfect sense to have the Volume be a control of the Codec driver even if it's provided by a separate chip on the same sound card? >> >> Not if they are standalone components that can also be used in different configurations on different sound cards. E.g. you code suggests that you are using a AK4113 and a DAC121C085. The CODEC driver is not the sound card driver, the sound card driver is the piece of code that glues all the different components that are found on the sound card together. >> >> i2c_new_dummy() is typically used for devices with multiple I2C addresses. Or where you have e.g. multiple devices with different addresses on the same die so they become a inseparable unit. >> >> - Lars > > Yes, but the functionality is very specific to the way the components are wired on the hardware. They are wired to work in conjunction with the ASIC Codec. > > The DAC121C085 is a very specific case as the DAC is actually wired to control the voltage of 4 quadrant multiplier which is hardwired to the Codec ASIC. This is a prototype. In the final product, the DAC chip may be built into the ASIC. For this one, I feel like the "Volume" should be a control of the same driver as the ASIC. > > This said I see your point about the AK4113 which is a generic multi channel receiver. In this specific application it is wired so only inputs 1,2 and 3 are hard wired (this actually isn't a sound card, but rather a finished product using Linux as the SoC OS as can be seen on the home page of http://birdland.com right now). > > There is already a working prototype, but as of now, the inputs are selected by the application talking directly to the I2C AKM. I'm just in the process of streamlining the code, and naturally, I envisioned the input selection be part of the sound card codec as I'd much rather have the sound card have a enum control with the 4 inputs shown by name show under amixer (this would enable my application to be more generic). > > Can you please point me to an example of a Sound board driver that uses multiple codecs? I suppose I can modify my sound board driver to use a codec for the audio and another for the AKM input selector). Anything that uses aux devs. E.g. samsung/speyside.c There is also this series https://lkml.org/lkml/2014/4/5/92 which adds DT support for aux devs. - Lars

1 0

[alsa-devel] [PATCH 00/44 v3] Enhancement of support for Firewire devices
by Takashi Sakamoto 08 Apr '14

08 Apr '14

This series is an update for my previous series: [alsa-devel] [GIT PULL][PATCH 00/39 v2] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073868.html Updates since previous pull-request: firewire-lib: - remove packet sorting - remove adding transfer delay to out-packets - fix a bug of MIDI capturing for several ports - add detecting discontinuity of packets - add a support for deferred transaction - add some flags for device's quirk related to AMDTP packets - improve return value of some AV/C general commands fireworks: - add a workaround for out-of-specification dbc meaning - add a workaround for AudioFire9 firmware - add a workaround for firmware version 5.5 bebob: - add some flags for device's quirk related to AMDTP packets - add a workaround for discontinuity of packets at bus reset - add workarounds for M-Audio spefial Firewire series This time I intentionally add no restriction of version for Fireworks because I confirmed that my model doesn't always transmit packets with disorder. I expect users to retry starging streams several times till detecting no discontinuity of packets. [alsa-devel] [PATCH 09/39] firewire-lib: Add sort function for transmitted packet http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074157.html I note that a workaround for M-Audio special Firewire series at [44/44] relates to Linux Firewire subsystem. Please follow: Generating bus-reset by kernel module in the other subsystem http://sourceforge.net/p/linux1394/mailman/linux1394-devel/thread/5326B651.… ---------------------------------------------------------------- The following changes since commit 2df6742f613840a0b0a1590fb28f7af5b058a673: ALSA: hda - verify pin:cvt connection on preparing a stream for Intel HDMI codec (2014-03-20 07:36:17 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to aa0564fdd013e5754b73ba89b50c722bbedbc4ac: firewire/bebob: Add a workaround for M-Audio special Firewire series (2014-03-21 19:35:04 +0900) ---------------------------------------------------------------- Takashi Sakamoto (44): firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add support for channel mapping firewire-lib: Restrict calling flush_context_completion() when context exists firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add support for deferred transaction firewire-lib: Add some AV/C general commands fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks/firewire-lib: Add a quirk for empty packet with TAG0 fireworks/firewire-lib: Add a quirk for the meaning of dbc fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets fireworks/firewire-libs: Add a quirk for fixed interval of reported dbc fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks/firewire-lib: Add a quirk for data blocks for MIDI in out-stream fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob/firewire-lib: Add a quirk for discontinuity at bus reset bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series firewire/bebob: Add a workaround for M-Audio special Firewire series drivers/firewire/core.h | 1 - include/linux/firewire.h | 3 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 668 ++++++++++++---- sound/firewire/amdtp.h | 124 ++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 470 ++++++++++++ sound/firewire/bebob/bebob.h | 257 +++++++ sound/firewire/bebob/bebob_command.c | 311 ++++++++ sound/firewire/bebob/bebob_focusrite.c | 287 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 832 ++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 168 ++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 201 +++++ sound/firewire/bebob/bebob_stream.c | 929 +++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 75 ++ sound/firewire/bebob/bebob_yamaha.c | 54 ++ sound/firewire/cmp.c | 203 +++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 2 +- sound/firewire/fcp.c | 192 ++++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 331 ++++++++ sound/firewire/fireworks/fireworks.h | 227 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 +++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ sound/firewire/fireworks/fireworks_stream.c | 347 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 48 +- 37 files changed, 8157 insertions(+), 258 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

6 77

[alsa-devel] Audio codec with multiple I2C devices
by Gilles 07 Apr '14

07 Apr '14

Folks, I'm writing an ALSA SoC codec which uses a 3 distinct I2C devices (a DAC, a CAG for volume and a input selector) all 3 at different I2C addresses. My existing code relies on DT to declare the primary I2C address (in this case the DAC) and it works. But I now need to add support for the extended functionality. Can someone point me to the best way to declare the second I2C address in the DT and make use of it in the driver? I know I could always hard code it and add a new i2c_new_device() in the probe() into the private data but I'm trying to make it a bit more DT friendly. Not having much luck on finding examples of this. I'm attaching the existing code code below if anyone wants to reference to snipets of it. As you can see, the set/get volume are declared as _EXT since the volume is controlled by a secondary GC I2C device. Similarily, the input selector talks to a third I2C device. Any help would be appreciated. Thanks! /* // ALSA SoC codec driver for Birdland Audio SDM-50 Digital Amplifier // Copyright (C) 2012 Birdland Audio - http://birdland.com // Author: Gilles Gameiro <gilles(a)whospot.com> // */ #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/gpio.h> #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/initval.h> #include <sound/tlv.h> #define DRV_NAME "sdm2x-audio-codec" #define BIRDLAND_MODEL_250 0 #define SDM2X_REG0_VOLUME 0 #define SDM2X_REG1_MUXCLK 1 #define SDM2X_INPUT_EXTERNAL 0x80 #define SDM2X_PLAYRATE_48000 0x08 #define SDM2X_RATES ( SNDRV_PCM_RATE_32000 \ | SNDRV_PCM_RATE_44100 \ | SNDRV_PCM_RATE_48000 \ | SNDRV_PCM_RATE_64000 \ | SNDRV_PCM_RATE_88200 \ | SNDRV_PCM_RATE_96000 \ | SNDRV_PCM_RATE_176400 \ | SNDRV_PCM_RATE_192000 ) #define SDM2X_FORMATS ( SNDRV_PCM_FMTBIT_S16_LE \ | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE \ | SNDRV_PCM_FMTBIT_S32_LE ) /* codec private data */ struct sdm2x_priv { struct snd_soc_codec *codec; enum snd_soc_control_type control_type; u16 model; unsigned int sysclk; u8 input_cache; // Caches the back panel input selected (Controlled by AK4113) u8 volume_cache; // Caches the current value of the volume (Controlled by DAC121C085) }; // FPGA I2C Slave IP is write only. Use register cache #define SDM2X_RCACHE_SIZE 4 static const u8 sdm2x_reg_cache[SDM2X_RCACHE_SIZE] = { 0x00, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000), 0x00, 0x00, }; static int sdm250_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->volume_cache; return 0; } static int sdm250_set_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int volume = ucontrol->value.integer.value[0] & 0x7f; sdm2x->volume_cache = volume; // TODO: Write it to GA return 0; } static int sdm250_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->input_cache; return 0; } static int sdm250_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int input = ucontrol->value.integer.value[0] & 0x7f; sdm2x->input_cache = volume; // TODO: Write it to AK4113 return 0; } /* General Volume -127dB to 0dB on 1dB steps */ static DECLARE_TLV_DB_SCALE(general_volume_tlv, -12700, 100, 0); /* Enum declarations */ static const char *sdm2x_srcin[] = {"Internal", "External"}; static const struct soc_enum sdm2x_srcin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 7, 2, sdm2x_srcin); static const char *sdm2x_i2sin[] = {"Optical 1", "Optical 2", "Optical 3", "XLR" }; static const struct soc_enum sdm2x_i2sin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 0, 3, sdm2x_i2sin); static const struct snd_kcontrol_new sdm2x_snd_controls[] = { /* General Volume: This is set outside of the FPGA so use getter/setter functions */ SOC_SINGLE_EXT_TLV("Amplifier Volume", SDM2X_REG0_VOLUME, 0, 127, 0, sdm250_get_volume, sdm250_set_volume, general_volume_tlv), /* FPGA Manged Controls (primary I2C of this driver) */ SOC_SINGLE("Digital Mute", SDM2X_REG0_VOLUME, 7, 1, 0), // TODO: Make this a DAI op? SOC_ENUM("SRC Input Source", sdm2x_srcin_enum), /* Back Pannel Controls (AK4113) */ SOC_ENUM_EXT("Back Panel Input", sdm2x_i2sin_enum, sdm250_get_input, sdm250_set_input) }; static int sdm2x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; //struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int fsref; unsigned char divreg; // Set Data word length not needed in I2S) fsref = params_format(params); // Set Sampling Rate: I haven't figured out how to make the mcasp I2S master for TX and slave for RX so let the FPGA generate the clocks Sampling Rate switch (fsref) { case SNDRV_PCM_RATE_32000: divreg = 0x0C; break; case SNDRV_PCM_RATE_44100: case SNDRV_PCM_RATE_48000: divreg = 0x08; break; case SNDRV_PCM_RATE_64000: divreg = 0x06; break; case SNDRV_PCM_RATE_88200: case SNDRV_PCM_RATE_96000: divreg = 0x04; break; case SNDRV_PCM_RATE_176400: case SNDRV_PCM_RATE_192000: divreg = 0x02; break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested rate 0x%08x not supported\n", __func__, fsref); return -EINVAL; } if (fsref % 11025 == 0) divreg |= 0x10; snd_soc_update_bits(codec, SDM2X_REG1_MUXCLK, 0x1F, divreg); return 0; } // DAI Ops static int sdm2x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { //struct snd_soc_codec *codec = codec_dai->codec; //struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // SDM-50 only supports vanilla I2S format. switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Format 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_FORMAT_MASK)); return -EINVAL; } // SDM-50 Uses normal I2S Clock polarities switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Clock Polarity 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_INV_MASK)); return -EINVAL; } // And SDM-50 is the I2S Clock Master for both Bit and Frame switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case (SND_SOC_DAIFMT_CBM_CFM): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Master/Slave Clock 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_MASTER_MASK)); return -EINVAL; } return 0; } static int sdm2x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->sysclk = freq; return 0; } static const struct snd_soc_dai_ops sdm2x_dai_ops = { .hw_params = sdm2x_hw_params, .set_fmt = sdm2x_set_dai_fmt, .set_sysclk = sdm2x_set_dai_sysclk, //.digital_mute = sdm2x_mute, }; static struct snd_soc_dai_driver sdm2x_dai = { .name = "sdm2x-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .ops = &sdm2x_dai_ops, .symmetric_rates = 1, }; static int sdm2x_suspend(struct snd_soc_codec *codec) { return 0; } static int sdm2x_resume(struct snd_soc_codec *codec) { return 0; } /* PROBE: Setup private data, initialize codec and attach it to ASoC */ static int sdm2x_probe(struct snd_soc_codec *codec) { int ret; struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->codec = codec; // This belongs in the core but not sure if 3.8.13 supports it already.. ret = snd_soc_codec_set_cache_io(codec, 8, 8, sdm2x->control_type); if (ret != 0) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O: %d\n", __func__, ret); return ret; } //codec->cache_only = 1; // Init... snd_soc_write(codec, SDM2X_REG0_VOLUME, 0); snd_soc_write(codec, SDM2X_REG1_MUXCLK, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000)); //snd_soc_update_bits(codec, SDM250_REG2_STATUS, SOFT_RESET, SOFT_RESET); // Auto Clears after reset complete (might not work with cached registers) // Add AMIXER controls (Volume, etc..) snd_soc_add_codec_controls(codec, sdm2x_snd_controls, ARRAY_SIZE(sdm2x_snd_controls)); return 0; } static int sdm2x_remove(struct snd_soc_codec *codec) { // struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // TODO: Where is private data freed up? return 0; } static struct snd_soc_codec_driver soc_codec_dev_sdm2x = { .reg_cache_size = ARRAY_SIZE(sdm2x_reg_cache), .reg_word_size = sizeof(u8), .reg_cache_default = sdm2x_reg_cache, .probe = sdm2x_probe, .remove = sdm2x_remove, .suspend = sdm2x_suspend, .resume = sdm2x_resume, }; static const struct i2c_device_id sdm2x_i2c_id[] = { { "sdm2x", BIRDLAND_MODEL_250 }, { } }; MODULE_DEVICE_TABLE(i2c, sdm2x_i2c_id); /* If the i2c layer weren't so broken, we could pass this kind of data around */ static int sdm2x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sdm2x_priv *sdm2x; struct device_node *np = i2c->dev.of_node; int ret; sdm2x = devm_kzalloc(&i2c->dev, sizeof(struct sdm2x_priv), GFP_KERNEL); if (sdm2x == NULL) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O\n", __func__); return -ENOMEM; } // This driver doesn't support the old style platform data if (!np) { printk(KERN_ERR "sdm2x-codec.%s(): missing DT data info\n", __func__); return -EINVAL; } sdm2x->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, sdm2x); sdm2x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_sdm2x, &sdm2x_dai, 1); return ret; } static int sdm2x_i2c_remove(struct i2c_client *client) { // TODO: Where is private data freed up? snd_soc_unregister_codec(&client->dev); return 0; } #if defined(CONFIG_OF) static const struct of_device_id sdm2x_of_match[] = { { .compatible = "birdland,sdm2x", }, {}, }; MODULE_DEVICE_TABLE(of, sdm2x_of_match); #endif /* machine i2c codec control layer */ static struct i2c_driver sdm2x_i2c_driver = { .driver = { .name = "sdm2x-codec", .owner = THIS_MODULE, .of_match_table = of_match_ptr(sdm2x_of_match), }, .probe = sdm2x_i2c_probe, .remove = sdm2x_i2c_remove, .id_table = sdm2x_i2c_id, }; module_i2c_driver(sdm2x_i2c_driver); MODULE_DESCRIPTION("ASoC SDM2x codec driver"); MODULE_AUTHOR("Gilles Gameiro"); MODULE_LICENSE("GPL");

3 3

[alsa-devel] [PATCH v9] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 07 Apr '14

07 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- Changes since v8: - module_usb_driver() macro for module initialization - improved some comments Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 461 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 478 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..820d6ca --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,461 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* return URB to device */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* pass URB to device to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +module_usb_driver(bcd2000_driver); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

4 5

[alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Eric F 07 Apr '14

07 Apr '14

Hello, Been searching for days on Internet for this issue, without luck :( I own a Teac UD-H01 : when connected via USB => crackles and clicks

4 7

[alsa-devel] [PATCH] ASoC: da732x: Print correct major id
by Axel Lin 07 Apr '14

07 Apr '14

DA732X_ID_MAJOR_MASK is 0xF0, so the major id is (reg & DA732X_ID_MAJOR_MASK) >> 4. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da732x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 7d168ec..48f3fef 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1571,7 +1571,8 @@ static int da732x_i2c_probe(struct i2c_client *i2c, } dev_info(&i2c->dev, "Revision: %d.%d\n", - (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + (reg & DA732X_ID_MAJOR_MASK) >> 4, + (reg & DA732X_ID_MINOR_MASK)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, da732x_dai, ARRAY_SIZE(da732x_dai)); -- 1.8.3.2

3 2

[alsa-devel] [PATCH alsa-lib] pcm: rate: Don't return negative frame count on success in rewind
by Andrew Eikum 07 Apr '14

07 Apr '14

snd_pcm_rewind is documented to return <0 on failure and >=0 on success. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- I checked all other rewind implementations, and this was the only one that obviously had this bug. diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 54a3e67..2eb4b1b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -702,7 +702,7 @@ static snd_pcm_sframes_t snd_pcm_rate_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f snd_atomic_write_begin(&rate->watom); n = snd_pcm_rate_move_applptr(pcm, -frames); snd_atomic_write_end(&rate->watom); - return n; + return n < 0 ? -n : n; } static snd_pcm_sframes_t snd_pcm_rate_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames) -- 1.9.1

3 2

[alsa-devel] [PATCH] Allow snd_soc_update_bits use 32 bits register
by bardliao＠realtek.com 07 Apr '14

07 Apr '14

From: Bard Liao <bardliao(a)realtek.com> Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..687dec0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,7 +469,7 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..fde636c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask for REVID
by Axel Lin 07 Apr '14

07 Apr '14

BIT[0:2] of register 01h is REVID, so the mask for REVID should be 0x7. Also updates the code to use CS42L52_CHIP_REV_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l52.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6be..460d355 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00..ac445993 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -37,7 +37,7 @@ #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: wm5110: Add in OSR controls for OUT5/6
by Charles Keepax 07 Apr '14

07 Apr '14

There are no OSR controls on outputs 1-4 on wm5110, however when these were removed the ones on output 5 and 6 were also accidentally removed, but those actually exist. This patch adds these controls back in. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index df5a38d..83a7e2f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -367,6 +367,11 @@ SOC_SINGLE("HPOUT2 SC Protect Switch", ARIZONA_HP2_SHORT_CIRCUIT_CTRL, SOC_SINGLE("HPOUT3 SC Protect Switch", ARIZONA_HP3_SHORT_CIRCUIT_CTRL, ARIZONA_HP3_SC_ENA_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, + ARIZONA_OUT6_OSR_SHIFT, 1, 0), + SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, -- 1.7.2.5

2 1

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 06 Apr '14

06 Apr '14

Continued from here: http://article.gmane.org/gmane.linux.alsa.devel/121379 After a few days now I finally got VLC to log everything that happens, and finally the bug reappeared, and unlike previously thought, it happens when I watch a video as well. But unfortunately VLC didn't give anything specific that happened at the time, just some vague ALSA errors about broken pipes (at line 394) : http://pastebin.com/NgZiG9yX At the time I added the video to VLC's queue I was doing something in the background (which I don't remember now unfortunately), but this has happened even when listening to something without doing anything in the background, and not having any background-running apps except Transmission. Also interestingly lately I've heard a few crackles/pops when listening to stuff, followed by a second or so of silence then back to normal when I would run CPU intensive stuff. At those points I thought the bug would be invoked again, but it didn't happen so. I still have the crackling problem, not able to change the PCM channel volume and screeching on audio that has clipped portions or very high frequency waves. Can anyone give me any instructions on what I could do to further diagnose this problem? And more importantly, as I stated in the linked post above, how could I shut down the sound daemon/server/whatever so I could bring back the OS to stability without restarting.

2 4

[alsa-devel] [PATCH] ASoC: cs42l52: Report correct chip id and revision
by Axel Lin 05 Apr '14

05 Apr '14

According to the datasheet: Chip I.D and Revision Register (Address 01h) BIT[0:2] REVID BIT[3:7] CHIPID (CS42L52: 11100) REVID takes 3 bits, so CS42L52_CHIP_REV_MASK should be 0x07. While at it, also adds define for CS42L52_CHIP_REV_B1. The CHIPID takes BIT[3:7], so this patch updates the defines and the code to show correct chip id for this chip. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 ++-- sound/soc/codecs/cs42l52.h | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6be..277df17 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1249,7 +1249,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, ret); ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg); - devid = reg & CS42L52_CHIP_ID_MASK; + devid = (reg & CS42L52_CHIP_ID_MASK) >> 3; if (devid != CS42L52_CHIP_ID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00..ce7ce5a 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -32,12 +32,13 @@ #define CS42L52_FIX_BITS_CTL 0x00 #define CS42L52_CHIP 0x01 -#define CS42L52_CHIP_ID 0xE0 -#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_ID 0x1C +#define CS42L52_CHIP_ID_MASK (0x1F << 3) #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_B1 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F -- 1.8.3.2

3 9

[alsa-devel] [PATCH] sound/oss: Remove uncompilable DBG macro use
by Joe Perches 04 Apr '14

04 Apr '14

Most of it duplicates function tracing and one of them has an uncompilable printf %P use. Others have format/argument mismatches. Remove unused DBG1 macro definition Neaten uart401.c use of ok test around this DBG macro removal. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/ad1848.c | 4 ---- sound/oss/opl3.c | 3 --- sound/oss/pas2_mixer.c | 9 --------- sound/oss/pas2_pcm.c | 18 ------------------ sound/oss/sb_common.c | 4 ---- sound/oss/sb_ess.c | 4 ---- sound/oss/sequencer.c | 6 ------ sound/oss/sound_config.h | 4 ---- sound/oss/soundcard.c | 6 ------ sound/oss/uart401.c | 11 ++--------- 10 files changed, 2 insertions(+), 67 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 4918b71..ec1ee07 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -50,8 +50,6 @@ #include <linux/pnp.h> #include <linux/spinlock.h> -#define DEB(x) -#define DEB1(x) #include "sound_config.h" #include "ad1848.h" @@ -1016,8 +1014,6 @@ static void ad1848_close(int dev) ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; - DEB(printk("ad1848_close(void)\n")); - devc->intr_active = 0; ad1848_halt(dev); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index c5c2440..4709e59 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -275,7 +275,6 @@ static int opl3_kill_note (int devno, int voice, int note, int velocity) devc->v_alloc->map[voice] = 0; map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Kill note %d\n", voice)); if (map->voice_mode == 0) return 0; @@ -873,8 +872,6 @@ static void opl3_aftertouch(int dev, int voice, int pressure) map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Aftertouch %d\n", voice)); - if (map->voice_mode == 0) return; diff --git a/sound/oss/pas2_mixer.c b/sound/oss/pas2_mixer.c index a0bcb85..50b5bd5 100644 --- a/sound/oss/pas2_mixer.c +++ b/sound/oss/pas2_mixer.c @@ -21,10 +21,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(what) /* (what) */ -#endif - extern int pas_translate_code; extern char pas_model; extern int *pas_osp; @@ -120,8 +116,6 @@ pas_mixer_set(int whichDev, unsigned int level) { int left, right, devmask, changed, i, mixer = 0; - DEB(printk("static int pas_mixer_set(int whichDev = %d, unsigned int level = %X)\n", whichDev, level)); - left = level & 0x7f; right = (level & 0x7f00) >> 8; @@ -207,8 +201,6 @@ pas_mixer_reset(void) { int foo; - DEB(printk("pas2_mixer.c: void pas_mixer_reset(void)\n")); - for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++) pas_mixer_set(foo, levels[foo]); @@ -220,7 +212,6 @@ static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) int level,v ; int __user *p = (int __user *)arg; - DEB(printk("pas2_mixer.c: int pas_mixer_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */ if (get_user(level, p)) return -EFAULT; diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c index 6f13ab4..474803b 100644 --- a/sound/oss/pas2_pcm.c +++ b/sound/oss/pas2_pcm.c @@ -22,10 +22,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(WHAT) -#endif - #define PAS_PCM_INTRBITS (0x08) /* * Sample buffer timer interrupt enable @@ -156,8 +152,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) int val, ret; int __user *p = arg; - DEB(printk("pas2_pcm.c: static int pas_audio_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); - switch (cmd) { case SOUND_PCM_WRITE_RATE: @@ -204,8 +198,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) static void pas_audio_reset(int dev) { - DEB(printk("pas2_pcm.c: static void pas_audio_reset(void)\n")); - pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */ } @@ -214,8 +206,6 @@ static int pas_audio_open(int dev, int mode) int err; unsigned long flags; - DEB(printk("pas2_pcm.c: static int pas_audio_open(int mode = %X)\n", mode)); - spin_lock_irqsave(&pas_lock, flags); if (pcm_busy) { @@ -239,8 +229,6 @@ static void pas_audio_close(int dev) { unsigned long flags; - DEB(printk("pas2_pcm.c: static void pas_audio_close(void)\n")); - spin_lock_irqsave(&pas_lock, flags); pas_audio_reset(dev); @@ -256,8 +244,6 @@ static void pas_audio_output_block(int dev, unsigned long buf, int count, { unsigned long flags, cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_output_block(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -303,8 +289,6 @@ static void pas_audio_start_input(int dev, unsigned long buf, int count, unsigned long flags; int cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_start_input(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -388,8 +372,6 @@ static struct audio_driver pas_audio_driver = void __init pas_pcm_init(struct address_info *hw_config) { - DEB(printk("pas2_pcm.c: long pas_pcm_init()\n")); - pcm_bitsok = 8; if (pas_read(0xEF8B) & 0x08) pcm_bitsok |= 16; diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 851a1da..3d50fb4 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -226,8 +226,6 @@ int sb_dsp_reset(sb_devc * devc) { int loopc; - DEB(printk("Entered sb_dsp_reset()\n")); - if (devc->model == MDL_ESS) return ess_dsp_reset (devc); /* This is only for non-ESS chips */ @@ -246,8 +244,6 @@ int sb_dsp_reset(sb_devc * devc) return 0; /* Sorry */ } - DEB(printk("sb_dsp_reset() OK\n")); - return 1; } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 0e7254b..b47a690 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -865,8 +865,6 @@ printk(KERN_INFO "FKS: ess_dsp_reset 1\n"); ess_show_mixerregs (devc); #endif - DEB(printk("Entered ess_dsp_reset()\n")); - outb(3, DSP_RESET); /* Reset FIFO too */ udelay(10); @@ -881,8 +879,6 @@ ess_show_mixerregs (devc); } ess_extended (devc); - DEB(printk("sb_dsp_reset() OK\n")); - #ifdef FKS_LOGGING printk(KERN_INFO "FKS: dsp_reset 2\n"); ess_show_mixerregs (devc); diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 9b9f7d3..c0eea1d 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -216,8 +216,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun dev = dev >> 4; - DEB(printk("sequencer_write(dev=%d, count=%d)\n", dev, count)); - if (mode == OPEN_READ) return -EIO; @@ -959,8 +957,6 @@ int sequencer_open(int dev, struct file *file) dev = dev >> 4; mode = translate_mode(file); - DEB(printk("sequencer_open(dev=%d)\n", dev)); - if (!sequencer_ok) { /* printk("Sound card: sequencer not initialized\n");*/ @@ -1133,8 +1129,6 @@ void sequencer_release(int dev, struct file *file) dev = dev >> 4; - DEB(printk("sequencer_release(dev=%d)\n", dev)); - /* * Wait until the queue is empty (if we don't have nonblock) */ diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 9d35c4c..f2554ab 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -123,10 +123,6 @@ static inline int translate_mode(struct file *file) #include "sound_calls.h" #include "dev_table.h" -#ifndef DEB -#define DEB(x) -#endif - #ifndef DDB #define DDB(x) do {} while (0) #endif diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index e778034..b70c7c8 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -154,7 +154,6 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof mutex_lock(&soundcard_mutex); - DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_DSP: case SND_DEV_DSP16: @@ -180,7 +179,6 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou int ret = -EINVAL; mutex_lock(&soundcard_mutex); - DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: @@ -206,7 +204,6 @@ static int sound_open(struct inode *inode, struct file *file) int dev = iminor(inode); int retval; - DEB(printk("sound_open(dev=%d)\n", dev)); if ((dev >= SND_NDEVS) || (dev < 0)) { printk(KERN_ERR "Invalid minor device %d\n", dev); return -ENXIO; @@ -257,7 +254,6 @@ static int sound_release(struct inode *inode, struct file *file) int dev = iminor(inode); mutex_lock(&soundcard_mutex); - DEB(printk("sound_release(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_CTL: module_put(mixer_devs[dev >> 4]->owner); @@ -351,7 +347,6 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (!access_ok(VERIFY_WRITE, p, len)) return -EFAULT; } - DEB(printk("sound_ioctl(dev=%d, cmd=0x%x, arg=0x%x)\n", dev, cmd, arg)); if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); @@ -409,7 +404,6 @@ static unsigned int sound_poll(struct file *file, poll_table * wait) struct inode *inode = file_inode(file); int dev = iminor(inode); - DEB(printk("sound_poll(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 5433c6f..62b8869 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -274,19 +274,12 @@ static int reset_uart401(uart401_devc * devc) } } - + /* Flush input before enabling interrupts */ if (ok) - { - DEB(printk("Reset UART401 OK\n")); - } + uart401_input_loop(devc); else DDB(printk("Reset UART401 failed - No hardware detected.\n")); - if (ok) - uart401_input_loop(devc); /* - * Flush input before enabling interrupts - */ - return ok; }

2 1

[alsa-devel] [PATCH 1/2] snd-ice1712: restore AK4xxx volumes on resume
by Ondrej Zary 04 Apr '14

04 Apr '14

Also restore AK4xxx mixer volumes on resume for M-Audio ICE1712-based cards. This fixes incorrect (sound working) zero mixer volumes after resume. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ice1712/delta.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 3764a27..bb34608 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -578,7 +578,8 @@ static struct snd_ak4xxx_private akm_vx442_priv = { #ifdef CONFIG_PM_SLEEP static int snd_ice1712_delta_resume(struct snd_ice1712 *ice) { - unsigned char akm_backup[AK4XXX_IMAGE_SIZE]; + unsigned char akm_img_bak[AK4XXX_IMAGE_SIZE]; + unsigned char akm_vol_bak[AK4XXX_IMAGE_SIZE]; /* init spdif */ switch (ice->eeprom.subvendor) { @@ -603,9 +604,11 @@ static int snd_ice1712_delta_resume(struct snd_ice1712 *ice) /* init codec and restore registers */ if (ice->akm_codecs) { - memcpy(akm_backup, ice->akm->images, sizeof(akm_backup)); + memcpy(akm_img_bak, ice->akm->images, sizeof(akm_img_bak)); + memcpy(akm_vol_bak, ice->akm->volumes, sizeof(akm_vol_bak)); snd_akm4xxx_init(ice->akm); - memcpy(ice->akm->images, akm_backup, sizeof(akm_backup)); + memcpy(ice->akm->images, akm_img_bak, sizeof(akm_img_bak)); + memcpy(ice->akm->volumes, akm_vol_bak, sizeof(akm_vol_bak)); snd_akm4xxx_reset(ice->akm, 0); } -- Ondrej Zary

2 2

[alsa-devel] [PATCH v2 00/16] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 04 Apr '14

04 Apr '14

Hi, This new version of the cleanup is mainly a reordering of the patches. It moves the regmap patch to the end of the series so none of the other patches depend on it. This also fixes a bug I introduced in "Remove fsl_ssi_setup" which also removed the switching between network and normal mode depending on the number of channels. I reversed this code removal. Another new patch was added "Fix i2s_mode variable setup" which fixes the i2s_mode assignment in set_dai_fmt(). Best regards, Markus Changes in v2: - Move "Use regmap" patch to the end of the series - Add normal/network mode switching code to again (previously removed in "Remove fsl_ssi_setup" - Add patch "Fix i2s_mode variable setup" Markus Pargmann (16): ASoC: fsl-ssi: Remove fsl_ssi_setup ASoC: fsl-ssi: Fix i2s_mode variable setup ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: baud clock error handling ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Only enable baudclk when used ASoC: fsl-ssi: Set default dai-fmts ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Update binding documentation ASoC: fsl-ssi: Use regmap .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 1173 +++++++++----------- sound/soc/fsl/fsl_ssi.h | 112 +- sound/soc/fsl/fsl_ssi_dbg.c | 163 +++ 6 files changed, 759 insertions(+), 713 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.9.0

6 33

[alsa-devel] [PATCH] ASoC: alc56(23|32): fix undefined return value of probing code
by Christoph Jaeger 04 Apr '14

04 Apr '14

Commit 5d6be5aa ("ASoC: codec: Simplify ASoC probe code.") left variable 'ret', whose value is returned, uninitialized. Since it is not used otherwise, remove it. Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/soc/codecs/alc5623.c | 3 +-- sound/soc/codecs/alc5632.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 09f7e77..f500905 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -902,7 +902,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; alc5623_reset(codec); @@ -961,7 +960,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6..3ee8d4e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: SSI needs to prepare clocks before enabling
by Ben Dooks 04 Apr '14

04 Apr '14

From: William Towle <william.towle(a)codethink.co.uk> The rsnd_ssi_hw_{stop|start} functions need to call clk_prepare_enable() and clk_disable_unprepare(), otherwise the following warning occurs on attempt to play sound: WARNING: CPU: 2 PID: 1176 at drivers/clk/clk.c:926 __clk_enable+0x30/0x9c() due to drivers/clk/clk.c: 926 if (WARN_ON(clk->prepare_count == 0)) 927 return -ESHUTDOWN; Signed-off-by: William Towle <william.towle(a)codethink.co.uk> [ben.dooks(a)codethink.co.uk: add more bug info to report] Tested-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4b8cf7c..1c07ef7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -250,7 +250,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, u32 cr; if (0 == ssi->usrcnt) { - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); if (rsnd_rdai_is_clk_master(rdai)) { if (rsnd_ssi_clk_from_parent(ssi)) @@ -309,7 +309,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, rsnd_ssi_master_clk_stop(ssi); } - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); } dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); -- 1.9.0

3 2

[alsa-devel] [PATCH] ASoC: sirf: Move the tx rx enable from port to codec, that will not need register sharing
by RongJun Ying 04 Apr '14

04 Apr '14

From: Rongjun Ying <rongjun.ying(a)csr.com> The port driver only used to register component and dmaengine pcm. Signed-off-by: Rongjun Ying <rongjun.ying(a)csr.com> --- sound/soc/codecs/sirf-audio-codec.c | 74 +++++++++++++++++++++--- sound/soc/codecs/sirf-audio-codec.h | 50 ++++++++++++++++ sound/soc/sirf/sirf-audio-port.c | 107 ----------------------------------- sound/soc/sirf/sirf-audio-port.h | 62 -------------------- 4 files changed, 116 insertions(+), 177 deletions(-) delete mode 100644 sound/soc/sirf/sirf-audio-port.h diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 58e7c1f..c5177bc 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -279,13 +279,63 @@ static const struct snd_soc_dapm_route sirf_audio_codec_map[] = { {"Mic input mode mux", "Differential", "MICIN1"}, }; +static void sirf_audio_codec_tx_enable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, IC_TX_ENABLE); +} + +static void sirf_audio_codec_tx_disable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, ~IC_TX_ENABLE); +} + +static void sirf_audio_codec_rx_enable(struct sirf_audio_codec *sirf_audio_codec, + int channels) +{ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET); + regmap_write(sirf_audio_codec->regmap, + AUDIO_PORT_IC_RXFIFO_INT_MSK, 0); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + if (channels == 1) + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO); + else + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO); +} + +static void sirf_audio_codec_rx_disable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO); +} + static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_codec *codec = dai->codec; - u32 val = 0; + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; /* * This is a workaround, When stop playback, @@ -295,20 +345,28 @@ static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (playback) { + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, 0); + sirf_audio_codec_tx_disable(sirf_audio_codec); + } else + sirf_audio_codec_rx_disable(sirf_audio_codec); break; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (playback) - val = IC_HSLEN | IC_HSREN; + if (playback) { + sirf_audio_codec_tx_enable(sirf_audio_codec); + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, IC_HSLEN | IC_HSREN); + } else + sirf_audio_codec_rx_enable(sirf_audio_codec, + substream->runtime->channels); break; default: return -EINVAL; } - if (playback) - snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, - IC_HSLEN | IC_HSREN, val); return 0; } @@ -392,7 +450,7 @@ static const struct regmap_config sirf_audio_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = AUDIO_IC_CODEC_CTRL3, + .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK, .cache_type = REGCACHE_NONE, }; diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h index d4c187b..ba1adc0 100644 --- a/sound/soc/codecs/sirf-audio-codec.h +++ b/sound/soc/codecs/sirf-audio-codec.h @@ -72,4 +72,54 @@ #define IC_RXPGAR 0x7B #define IC_RXPGAL 0x7B +#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F +#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_HC_OFFSET) + +#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F +#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_HC_OFFSET) +#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) +#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) + +#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) +#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) +#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) +#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) +#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) + +#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) +#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) +#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) +#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) +#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) + +#define AUDIO_FIFO_START (1 << 0) +#define AUDIO_FIFO_RESET (1 << 1) + +#define AUDIO_FIFO_FULL (1 << 0) +#define AUDIO_FIFO_EMPTY (1 << 1) +#define AUDIO_FIFO_OFLOW (1 << 2) +#define AUDIO_FIFO_UFLOW (1 << 3) + +#define IC_TX_ENABLE (0x03) +#define IC_RX_ENABLE_MONO (0x01) +#define IC_RX_ENABLE_STEREO (0x03) + #endif /*__SIRF_AUDIO_CODEC_H*/ diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c index b04a53f..b4afa31 100644 --- a/sound/soc/sirf/sirf-audio-port.c +++ b/sound/soc/sirf/sirf-audio-port.c @@ -6,60 +6,15 @@ * Licensed under GPLv2 or later. */ #include <linux/module.h> -#include <linux/io.h> -#include <linux/regmap.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include "sirf-audio-port.h" - struct sirf_audio_port { struct regmap *regmap; struct snd_dmaengine_dai_dma_data playback_dma_data; struct snd_dmaengine_dai_dma_data capture_dma_data; }; -static void sirf_audio_port_tx_enable(struct sirf_audio_port *port) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, - AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0); - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, - AUDIO_FIFO_START, AUDIO_FIFO_START); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, - IC_TX_ENABLE, IC_TX_ENABLE); -} - -static void sirf_audio_port_tx_disable(struct sirf_audio_port *port) -{ - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, - IC_TX_ENABLE, ~IC_TX_ENABLE); -} - -static void sirf_audio_port_rx_enable(struct sirf_audio_port *port, - int channels) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, - AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); - regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0); - regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, - AUDIO_FIFO_START, AUDIO_FIFO_START); - if (channels == 1) - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO); - else - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO); -} - -static void sirf_audio_port_rx_disable(struct sirf_audio_port *port) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO); -} static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) { @@ -69,41 +24,6 @@ static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) return 0; } -static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai); - int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (playback) - sirf_audio_port_tx_disable(port); - else - sirf_audio_port_rx_disable(port); - break; - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (playback) - sirf_audio_port_tx_enable(port); - else - sirf_audio_port_rx_enable(port, - substream->runtime->channels); - break; - default: - return -EINVAL; - } - - return 0; -} - -static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = { - .trigger = sirf_audio_port_trigger, -}; - static struct snd_soc_dai_driver sirf_audio_port_dai = { .probe = sirf_audio_port_dai_probe, .name = "sirf-audio-port", @@ -120,49 +40,22 @@ static struct snd_soc_dai_driver sirf_audio_port_dai = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &sirf_audio_port_dai_ops, }; static const struct snd_soc_component_driver sirf_audio_port_component = { .name = "sirf-audio-port", }; -static const struct regmap_config sirf_audio_port_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK, - .cache_type = REGCACHE_NONE, -}; - static int sirf_audio_port_probe(struct platform_device *pdev) { int ret; struct sirf_audio_port *port; - void __iomem *base; - struct resource *mem_res; port = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_port), GFP_KERNEL); if (!port) return -ENOMEM; - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; - } - - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; - - port->regmap = devm_regmap_init_mmio(&pdev->dev, base, - &sirf_audio_port_regmap_config); - if (IS_ERR(port->regmap)) - return PTR_ERR(port->regmap); - ret = devm_snd_soc_register_component(&pdev->dev, &sirf_audio_port_component, &sirf_audio_port_dai, 1); if (ret) diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h deleted file mode 100644 index f32dc54..0000000 --- a/sound/soc/sirf/sirf-audio-port.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * SiRF Audio port controllers define - * - * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. - * - * Licensed under GPLv2 or later. - */ - -#ifndef _SIRF_AUDIO_PORT_H -#define _SIRF_AUDIO_PORT_H - -#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F -#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 - -#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_SC_OFFSET) -#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_LC_OFFSET) -#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_HC_OFFSET) - -#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F -#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 - -#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_SC_OFFSET) -#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_LC_OFFSET) -#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_HC_OFFSET) -#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) -#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) - -#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) -#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) -#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) -#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) -#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) - -#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) -#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) -#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) -#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) -#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) - -#define AUDIO_FIFO_START (1 << 0) -#define AUDIO_FIFO_RESET (1 << 1) - -#define AUDIO_FIFO_FULL (1 << 0) -#define AUDIO_FIFO_EMPTY (1 << 1) -#define AUDIO_FIFO_OFLOW (1 << 2) -#define AUDIO_FIFO_UFLOW (1 << 3) - -#define IC_TX_ENABLE (0x03) -#define IC_RX_ENABLE_MONO (0x01) -#define IC_RX_ENABLE_STEREO (0x03) - -#endif /*__SIRF_AUDIO_PORT_H*/ -- 1.7.5.4

2 3

[alsa-devel] [PATCH v2] ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
by Axel Lin 04 Apr '14

04 Apr '14

Fix checking return value of regmap_read(). Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0, so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Paul Handrigan <paul.handrigan(a)cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42xx8.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 082299a..8502032 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -495,17 +495,16 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) regcache_cache_bypass(cs42xx8->regmap, true); /* Validate the chip ID */ - regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); - if (val < 0) { - dev_err(dev, "failed to get device ID: %x", val); - ret = -EINVAL; + ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (ret < 0) { + dev_err(dev, "failed to get device ID, ret = %d", ret); goto err_enable; } /* The top four bits of the chip ID should be 0000 */ - if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + if (((val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4) != 0x00) { dev_err(dev, "unmatched chip ID: %d\n", - val & CS42XX8_CHIPID_CHIP_ID_MASK); + (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4); ret = -EINVAL; goto err_enable; } -- 1.8.3.2

3 2

[alsa-devel] [PATCH 0/2] Add ipg clock control to sai driver
by Nicolin Chen 04 Apr '14

04 Apr '14

This series of patches add ipg clock control to fsl_sai driver and updates the vf610.dtsi accordingly. @Shawn I'm not sure if VF610 currently does full works with SAI audio function. The PATCH-2 is based on broonie/for-next. Nicolin Chen (2): ASoC: fsl_sai: Add clock control for SAI ARM: dts: Add ipg clock for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 7 ++-- arch/arm/boot/dts/vf610.dtsi | 4 +-- sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 4 files changed, 43 insertions(+), 7 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 1/3] ASoC: alc5632: add an of_match table
by Stephen Warren 04 Apr '14

04 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> Add a device tree match table. This serves to make the driver's support of device tree more explicit. Perhaps the fallback for DT matching to using the i2c_device_id table will go away one day, since it fails in face of devices from different vendors with the same name. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 3ee8d4e41a99..85942ca36cbf 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1190,11 +1190,18 @@ static const struct i2c_device_id alc5632_i2c_table[] = { }; MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); +static const struct of_device_id alc5632_of_match[] = { + { .compatible = "realtek,alc5632", }, + { } +}; +MODULE_DEVICE_TABLE(of, alc5632_of_match); + /* i2c codec control layer */ static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(alc5632_of_match), }, .probe = alc5632_i2c_probe, .remove = alc5632_i2c_remove, -- 1.8.1.5

4 13

[alsa-devel] [PATCH] ASoC: dapm: Add support for multi register mux
by Arun Shamanna Lakshmi 04 Apr '14

04 Apr '14

- Modify soc_enum struct to handle pointers for reg and mask - Add dapm get and put APIs for multi register mux with one hot encoding - Update snd_soc_dapm_update struct to support multiple reg update Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 17 ++++- include/sound/soc.h | 34 +++++++-- sound/soc/soc-core.c | 12 ++-- sound/soc/soc-dapm.c | 174 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 197 insertions(+), 40 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..ded46732 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_ONEHOT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_onehot, \ + .put = snd_soc_dapm_put_enum_onehot, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -378,6 +384,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +600,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[3]; + int mask[3]; + int val[3]; + int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..add326a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,24 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1, .type = SND_SOC_ENUM_NONE } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1, .type = SND_SOC_ENUM_NONE } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, \ + .type = SND_SOC_ENUM_ONEHOT } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +299,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -326,6 +335,17 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +/* + * Soc Enum Type + * + * @NONE: soc_enum type for SINGLE, DOUBLE or VIRTUAL mux + * @ONEHOT: soc_enum type for one hot encoding mux + */ +enum snd_soc_enum_type { + SND_SOC_ENUM_NONE = 0, + SND_SOC_ENUM_ONEHOT = 1, +}; + struct device_node; struct snd_jack; struct snd_soc_card; @@ -1098,13 +1118,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + enum snd_soc_enum_type type; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..cf29722 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..19b004a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -511,13 +511,26 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, const struct snd_kcontrol_new *kcontrol) { struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; + unsigned int val, item, bit_pos = -1; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); + if (e->reg[0] != SND_SOC_NOPM) { + if (e->type == SND_SOC_ENUM_ONEHOT) { + for (i = 0; i < e->num_regs; i++) { + soc_widget_read(dest, e->reg[i], &val); + val = val & e->mask[i]; + if (val != 0) { + bit_pos = __ffs(val) + + (8 * dest->codec->val_bytes * i); + break; + } + } + item = snd_soc_enum_val_to_item(e, bit_pos); + } else { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; + item = snd_soc_enum_val_to_item(e, val); + } } else { /* since a virtual mux has no backing registers to * decide which path to connect, it will try to match @@ -1553,8 +1566,8 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; - int ret; + unsigned int wi, i; + int ret = 0; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; @@ -1575,8 +1588,12 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); + /* dapm update for multiple registers */ + for (i = 0; i < update->num_regs; i++) { + ret |= soc_widget_update_bits_locked(w, update->reg[i], + update->mask[i], update->val[i]); + } + if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2866,10 +2883,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; - + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2920,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2962,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +3002,112 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_onehot - dapm enumerated onehot mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = -1, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = __ffs(val) + (8 * codec->val_bytes * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_onehot); + +/** + * snd_soc_dapm_put_enum_onehot - dapm enumerated onehot mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + struct snd_soc_dapm_update update; + int ret = 0, reg_val = 0, i, update_idx = 0; + + if (item[0] >= e->items) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value >= 0) { + /* get the register index and value to set */ + reg_idx = value / (8 * codec->val_bytes); + bit_pos = value % (8 * codec->val_bytes); + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change = snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + /* set the selected register */ + update.reg[e->num_regs - 1] = e->reg[reg_idx]; + update.mask[e->num_regs - 1] = e->mask[reg_idx]; + update.val[e->num_regs - 1] = reg_val; + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + update.reg[update_idx] = e->reg[i]; + update.mask[update_idx] = e->mask[i]; + update.val[update_idx++] = 0; + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.num_regs = 3; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_onehot); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 11

[alsa-devel] [PATCH 0/4] Add ESAI & SPDIF Vybird and LS1+ support.
by Xiubo Li 04 Apr '14

04 Apr '14

Xiubo Li (4): ASoC: esai: fix buggy of 'ret' type definition. ASoC: esai: Add Vybird and LS1+ compatibles support. ASoC: spdif: Sort the header files alphabetically. ASoC: spdif: Add Vybird and LS1+ compatibles support. sound/soc/fsl/fsl_esai.c | 10 +++++++--- sound/soc/fsl/fsl_spdif.c | 10 ++++++---- 2 files changed, 13 insertions(+), 7 deletions(-) -- 1.8.4

4 12

[alsa-devel] [PATCH RESEND 1/2] ASoC: fsl: Separation of the main audio options and boards
by Alexander Shiyan 04 Apr '14

04 Apr '14

This patch provide separation of the main audio options and boards. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 597962e..94c9350 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,5 @@ +menu "SoC Audio for Freescale CPUs" + config SND_SOC_FSL_SAI tristate select REGMAP_MMIO @@ -17,13 +19,27 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate -menuconfig SND_POWERPC_SOC +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. +config SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + if SND_POWERPC_SOC config SND_MPC52xx_DMA @@ -32,6 +48,8 @@ config SND_MPC52xx_DMA config SND_SOC_POWERPC_DMA tristate +comment "SoC Audio support for Freescale PPC boards:" + config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver @@ -109,13 +127,6 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - if SND_IMX_SOC config SND_SOC_IMX_SSI @@ -125,12 +136,7 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate +comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" @@ -223,3 +229,5 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +endmenu -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
by Axel Lin 03 Apr '14

03 Apr '14

Fix checking return value of regmap_read(). Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0, so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42xx8.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 082299a..b1606cf 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -495,17 +495,16 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) regcache_cache_bypass(cs42xx8->regmap, true); /* Validate the chip ID */ - regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); - if (val < 0) { + ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (ret < 0) { dev_err(dev, "failed to get device ID: %x", val); - ret = -EINVAL; goto err_enable; } /* The top four bits of the chip ID should be 0000 */ - if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + if (((val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4) != 0x00) { dev_err(dev, "unmatched chip ID: %d\n", - val & CS42XX8_CHIPID_CHIP_ID_MASK); + (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4); ret = -EINVAL; goto err_enable; } -- 1.8.3.2

5 4

Re: [alsa-devel] [Intel-gfx] [PATCH v2] drm/i915: move power domain init earlier during system resume
by Daniel Vetter 03 Apr '14

03 Apr '14

On Tue, Apr 01, 2014 at 09:50:43PM +0300, Imre Deak wrote: > On Tue, 2014-04-01 at 19:48 +0200, Daniel Vetter wrote: > > On Tue, Apr 01, 2014 at 07:55:22PM +0300, Imre Deak wrote: > > > During resume the intel hda audio driver depends on the i915 driver > > > reinitializing the audio power domain. Since the order of calling the > > > i915 resume handler wrt. that of the audio driver is not guaranteed, > > > move the power domain reinitialization step to the resume_early > > > handler. This is guaranteed to run before the resume handler of any > > > other driver. > > > > > > The power domain initialization in turn requires us to enable the i915 > > > pci device first, so move that part earlier too. > > > > > > Accordingly disabling of the i915 pci device should happen after the > > > audio suspend handler ran. So move the disabling later from the i915 > > > resume handler to the resume_late handler. > > > > > > v2: > > > - move intel_uncore_sanitize/early_sanitize earlier too, so they don't > > > get reordered wrt. intel_power_domains_init_hw() > > > > > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=76152 > > > Signed-off-by: Imre Deak <imre.deak(a)intel.com> > > > > So this is kinda why we should have gone with something proper, like a new > > hdmi sink platform device created by i915 and registered as a driver by > > snd-hda. Then the power domains stuff in the device core should take care > > of these kinds of ordering issues. Or at least snd-hda can tell it that it > > needs to wait for the hdmi-sink power domain to go on first before it can > > resume, I'm not really fluent on the details here. > > > > And having a hdmi sink bus would allow us to throw all kinds of crap into > > a clearly-defined interface, e.g. eld handling, hdcp synchronization, hpd > > forwarding and all the other fun stuff. > > > > So not sure what I should do with this here now. > > Right, I'm not too happy about this solution either, so if anything it > could be considered only a stop-gap fix. What you suggest seems to be a > cleaner way but it'd require more time to investigate/implement at least > on my part (but I'm ok to put it on my TODO list). We'd definitely need to discuss this with Takashi Iwai, since it would only really be useful if it's good enough to solve the general pile of sound/gfx coordination issues we have with hdmi. Adding him and alsa-dev. -Daniel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

2 4

[alsa-devel] [PATCH 1/2] snd-cs8427: separate HW initialization
by Ondrej Zary 03 Apr '14

03 Apr '14

Separate HW initialization from device creation. This is needed for suspend/resume support. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/cs8427.h | 1 + sound/i2c/cs8427.c | 57 ++++++++++++++++++++++++++++++++---------------- 2 files changed, 39 insertions(+), 19 deletions(-) diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h index f862cff..0b6a187 100644 --- a/include/sound/cs8427.h +++ b/include/sound/cs8427.h @@ -188,6 +188,7 @@ struct snd_pcm_substream; +int snd_cs8427_init(struct snd_i2c_bus *bus, struct snd_i2c_device *device); int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr, unsigned int reset_timeout, struct snd_i2c_device **r_cs8427); int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 6c2dc38..d9c15d9 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -150,10 +150,8 @@ static void snd_cs8427_free(struct snd_i2c_device *device) kfree(device->private_data); } -int snd_cs8427_create(struct snd_i2c_bus *bus, - unsigned char addr, - unsigned int reset_timeout, - struct snd_i2c_device **r_cs8427) +int snd_cs8427_init(struct snd_i2c_bus *bus, + struct snd_i2c_device *device) { static unsigned char initvals1[] = { CS8427_REG_CONTROL1 | CS8427_REG_AUTOINC, @@ -200,22 +198,10 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, Inhibit E->F transfers. */ CS8427_UD | CS8427_EFTUI | CS8427_DETUI, }; + struct cs8427 *chip = device->private_data; int err; - struct cs8427 *chip; - struct snd_i2c_device *device; unsigned char buf[24]; - if ((err = snd_i2c_device_create(bus, "CS8427", - CS8427_ADDR | (addr & 7), - &device)) < 0) - return err; - chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - snd_i2c_device_free(device); - return -ENOMEM; - } - device->private_free = snd_cs8427_free; - snd_i2c_lock(bus); err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); if (err != CS8427_VER8427A) { @@ -264,10 +250,44 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, snd_i2c_unlock(bus); /* turn on run bit and rock'n'roll */ + snd_cs8427_reset(device); + + return 0; + + __fail: + snd_i2c_unlock(bus); + + return err; +} + +EXPORT_SYMBOL(snd_cs8427_init); + +int snd_cs8427_create(struct snd_i2c_bus *bus, + unsigned char addr, + unsigned int reset_timeout, + struct snd_i2c_device **r_cs8427) +{ + int err; + struct cs8427 *chip; + struct snd_i2c_device *device; + + if ((err = snd_i2c_device_create(bus, "CS8427", + CS8427_ADDR | (addr & 7), + &device)) < 0) + return err; + chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + snd_i2c_device_free(device); + return -ENOMEM; + } + device->private_free = snd_cs8427_free; + if (reset_timeout < 1) reset_timeout = 1; chip->reset_timeout = reset_timeout; - snd_cs8427_reset(device); + + if ((err = snd_cs8427_init(bus, device))) + goto __fail; #if 0 // it's nice for read tests { @@ -286,7 +306,6 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, return 0; __fail: - snd_i2c_unlock(bus); snd_i2c_device_free(device); return err < 0 ? err : -EIO; } -- Ondrej Zary

2 6

[alsa-devel] [PATCH] ASoC: codec: fix the sparse check warnings.
by Xiubo Li 03 Apr '14

03 Apr '14

Some thing Likes: reproduce: make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/wm8997.c:1084:15: sparse: symbol \ 'wm8997_get_regmap' was not declared. Should it be static? Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index dc9e6b9..f073b6f 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1358,7 +1358,7 @@ static int pm860x_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *pm860x_get_regmap(struct device *dev) +static struct regmap *pm860x_get_regmap(struct device *dev) { struct pm860x_priv *pm860x = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 5e5518d..5ee48c8 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -152,7 +152,7 @@ static int cq93vc_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *cq93vc_get_regmap(struct device *dev) +static struct regmap *cq93vc_get_regmap(struct device *dev) { struct davinci_vc *davinci_vc = codec->dev->platform_data; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8c0cec3..9029e20 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -727,7 +727,7 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { } }; -struct regmap *mc13783_get_regmap(struct device *dev) +static struct regmap *mc13783_get_regmap(struct device *dev) { return dev_get_regmap(dev->parent, NULL); } diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 961b7e8..f26befb 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -231,7 +231,7 @@ static struct snd_soc_dai_driver si476x_dai = { .ops = &si476x_dai_ops, }; -struct regmap *si476x_get_regmap(struct device *dev) +static struct regmap *si476x_get_regmap(struct device *dev) { return dev_get_regmap(dev->parent, NULL); } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa60ba2..7a04653 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1798,7 +1798,7 @@ static unsigned int wm5102_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_5R, }; -struct regmap *wm5102_get_regmap(struct device *dev) +static struct regmap *wm5102_get_regmap(struct device *dev) { struct wm5102_priv *priv = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4be5f99..97eb1bc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1629,7 +1629,7 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_6R, }; -struct regmap *wm5110_get_regmap(struct device *dev) +static struct regmap *wm5110_get_regmap(struct device *dev) { struct wm5110_priv *priv = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 6b31a9f..1bd14b6 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1606,7 +1606,7 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8350_get_regmap(struct device *dev) +static struct regmap *wm8350_get_regmap(struct device *dev) { struct wm8350 *wm8350 = dev_get_platdata(dev); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index e6410f2..5880d22 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1359,7 +1359,7 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8400_get_regmap(struct device *dev) +static struct regmap *wm8400_get_regmap(struct device *dev) { struct wm8400 *wm8400 = dev_get_platdata(dev); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index daa4edc..6f336da 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4432,7 +4432,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8994_get_regmap(struct device *dev) +static struct regmap *wm8994_get_regmap(struct device *dev) { struct wm8994 *control = dev_get_drvdata(dev->parent); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 3d50621..09c4150 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1081,7 +1081,7 @@ static unsigned int wm8997_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_5R, }; -struct regmap *wm8997_get_regmap(struct device *dev) +static struct regmap *wm8997_get_regmap(struct device *dev) { struct wm8997_priv *priv = dev_get_drvdata(dev); -- 1.8.4

2 1

[alsa-devel] [PATCH] src/pcm/pcm_ladspa.c: Delay LADSPA plugin activate call
by Matthias Larisch 03 Apr '14

03 Apr '14

Some LADSPA Plugins rely on connected control ports on activate call. While this is not okay by spec, the spec also encourages the activate call happening as late as possible. Signed-off-by: Matthias Larisch <mail(a)matthias-larisch.de> --- src/pcm/pcm_ladspa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 9ce5242..7d1e3df 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -645,8 +645,6 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l return -EINVAL; } list_add_tail(&instance->list, &plugin->instances); - if (plugin->desc->activate) - plugin->desc->activate(instance->handle); if (plugin->policy == SND_PCM_LADSPA_POLICY_DUPLICATE) { err = snd_pcm_ladspa_connect_plugin_duplicate(plugin, &plugin->input, &plugin->output, instance, idx); if (err < 0) { @@ -664,6 +662,8 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l assert(err >= 0); err = snd_pcm_ladspa_connect_controls(plugin, &plugin->output, instance); assert(err >= 0); + if (plugin->desc->activate) + plugin->desc->activate(instance->handle); } err = snd_pcm_ladspa_check_connect(plugin, &plugin->input, &instance->input, depth); if (err < 0) -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix silent speaker output due to mute LED fixup
by Takashi Iwai 03 Apr '14

03 Apr '14

The recent fixups for HP laptops to support the mute LED made the speaker output silent on some machines. It turned out that they use the NID 0x18 for the speaker while it's also used for controlling the LED via VREF bits although the current driver code blindly assumes that such a node is a mic pin (where 0x18 is usually so). This patch fixes the problem by only changing the VREF bits and keeping the other pin ctl bits. Reported-and-tested-by: Hui Wang <hui.wang(a)canonical.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dba297288398..053107786f33 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3371,8 +3371,9 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) if (spec->mute_led_polarity) enabled = !enabled; - pinval = AC_PINCTL_IN_EN | - (enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid); + pinval &= ~AC_PINCTL_VREFEN; + pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80; if (spec->mute_led_nid) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } -- 1.9.1

1 0

[alsa-devel] Fixed single output empty pin_sense issue for alc283
by Kailang 03 Apr '14

03 Apr '14

Hi Takashi, Some machine with single output. It has only one HP out on it. alc283_init and alc283_shutup will get empty hp_pin value. I add the check for the attachment patch. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Add imx6sx platform support
by Nicolin Chen 03 Apr '14

03 Apr '14

The next coming i.MX6 Solo X SoC also contains SAI module while we use imp_pcm_init() for i.MX platform. So this patch adds one compatible route for imx6sx and updates the DT doc accordingly. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 12 ++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 98611a6..35c09fe 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,7 +7,7 @@ codec/DSP interfaces. Required properties: -- compatible: Compatible list, contains "fsl,vf610-sai". +- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. - clock-names : Must include the "sai" entry. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d64c33f..9ed6795 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -22,6 +22,7 @@ #include <sound/pcm_params.h> #include "fsl_sai.h" +#include "imx-pcm.h" #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ FSL_SAI_CSR_FEIE) @@ -592,6 +593,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + sai->sai_on_imx = true; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -634,12 +638,16 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (sai->sai_on_imx) + return imx_pcm_dma_init(pdev); + else + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, + { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index be26d46..677670d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -130,6 +130,7 @@ struct fsl_sai { bool big_endian_regs; bool big_endian_data; bool is_dsp_mode; + bool sai_on_imx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; -- 1.8.4

2 2

[alsa-devel] [PATCH v3] ASoC: sta350: Add codec driver
by Daniel Mack 02 Apr '14

02 Apr '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v2 -> v3: * Renamed the GPIO DT properties to *-gpios * Use DECLARE_TLV_DB_RANGE for automatic TLV_DB_RANGE_HEAD calculation * Use the new gpiod API * Some more cleanups v1 -> v2: * Added my S-o-b * Made the Kconfig symbol visible * Documented regulator notes in DT bindings doc * Fixed spelling and other style nits * Converted to SOC_ENUM_SINGLE_DECL * Added a lock around the logic in sta350_coefficient_get() * Added a DAPM route from "DAC" to "Playback" * Use dev_dbg() rather than pr_debug() * Changed the code to make use of regmap_update_bits rather than open-coding the same functionality * Cleaned up the reset sequence logic * If a gpio was requested, the claiming has to succeed (handles EPROBE_DEFER cases) * Kick codec->control_data initialization * Skip initialization of default values in regcache * Apply power before running init sequences * Parse the DT without consulting of_match_device() .../devicetree/bindings/sound/st,sta350.txt | 107 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1266 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1660 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..9501888 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,107 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * 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. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..c7b853f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,10 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate "STA350 speaker amplifier" + depends on I2C + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..552e92a --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1266 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.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. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.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 <sound/tlv.h> + +#include <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char * const sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + + struct gpio_desc *gpiod_nreset; + struct gpio_desc *gpiod_power_down; + + struct mutex coeff_lock; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char * const sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char * const sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char * const sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char * const sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char * const sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char * const sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char * const sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char * const sta350_noise_shaper_type[] = { + "Third order", "Fourth order" +}; + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +); + +static SOC_ENUM_SINGLE_DECL(sta350_drc_ac_enum, + STA350_CONFD, STA350_CONFD_DRC_SHIFT, + sta350_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta350_noise_shaper_enum, + STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + sta350_noise_shaper_type); +static SOC_ENUM_SINGLE_DECL(sta350_auto_gc_enum, + STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + sta350_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta350_auto_xo_enum, + STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + sta350_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch1_enum, + STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch2_enum, + STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch3_enum, + STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch1_enum, + STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch2_enum, + STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch3_enum, + STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_attack_rate_enum, + STA350_L1AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_attack_rate_enum, + STA350_L2AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_release_rate_enum, + STA350_L1AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_release_rate_enum, + STA350_L2AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i, ret = 0; + + mutex_lock(&sta350->coeff_lock); + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + } else if (numcoef == 5) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + } else { + ret = -EINVAL; + goto exit_unlock; + } + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + +exit_unlock: + mutex_unlock(&sta350->coeff_lock); + + return ret; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, + { "DAC", NULL, "Playback" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_C1IM | STA350_CONFB_C2IM, confb); +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + int ret; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + dev_dbg(codec->dev, "rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + } + + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) { + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + } + + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + confa = (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + confb = 0; + + switch (params_width(params)) { + case 24: + dev_dbg(codec->dev, "24bit\n"); + /* fall through */ + case 32: + dev_dbg(codec->dev, "24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case 20: + dev_dbg(codec->dev, "20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case 18: + dev_dbg(codec->dev, "18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case 16: + dev_dbg(codec->dev, "16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK, + confa); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB, + confb); + if (ret < 0) + return ret; + + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 1); + + if (sta350->gpiod_nreset) { + gpiod_set_value(sta350->gpiod_nreset, 0); + mdelay(1); + gpiod_set_value(sta350->gpiod_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + + /* power down: low */ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 0); + + if (sta350->gpiod_nreset) + gpiod_set_value(sta350->gpiod_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char * const sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(dev, sizeof(struct sta350_priv), GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + mutex_init(&sta350->coeff_lock); + sta350->pdata = dev_get_platdata(dev); + +#ifdef CONFIG_OF + if (dev->of_node) { + ret = sta350_probe_dt(dev, sta350); + if (ret < 0) + return ret; + } +#endif + + /* GPIOs */ + sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); + if (IS_ERR(sta350->gpiod_nreset)) { + ret = PTR_ERR(sta350->gpiod_nreset); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_nreset = NULL; + } else { + gpiod_direction_output(sta350->gpiod_nreset, 0); + } + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); + if (IS_ERR(sta350->gpiod_power_down)) { + ret = PTR_ERR(sta350->gpiod_power_down); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_power_down = NULL; + } else { + gpiod_direction_output(sta350->gpiod_power_down, 0); + } + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.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. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: Dynamic AFIFO and DMA configuration
by Peter Ujfalusi 02 Apr '14

02 Apr '14

Hi, Apart from the first patch which is a rebased resend of a previous patch this series will change the way McASP driver will configure the AFIFO and thus the DMA burst. The AFIFO and DMA configuration will be in one function so it is going to be easier to see the code and debug if needed. The main change is that instead of static tx/rx numevt (which is provided via DT or pdata) we are going to switch to dynamic configuration. With this setup we do not need to place any constraint on the period or buffer size since the AFIFO numevt and the corresponding DMA burst size will be picked according to period size, active serializers and the preferred numevt level. The only case when the code can fail if the period size in words can not be divided evenly by the active serializers, which is unlikely to happen. Regards, Peter --- Peter Ujfalusi (6): ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-mcasp: Fix debug typo in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Simplify and clean up the AFIFO configuration code ASoC: davinci-mcasp: Configure the AFIFO and DMA burst size at the same place ASoC: davinic-mcasp: Adopt the AFIFO/DMA configuration to the stream (dynamic depth) ASoC: davinci-mcasp: Fine tune and correct the DMA burst configuration sound/soc/davinci/davinci-mcasp.c | 145 +++++++++++++++++++++++--------------- sound/soc/davinci/davinci-mcasp.h | 1 + 2 files changed, 89 insertions(+), 57 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCHv2 00/11] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 02 Apr '14

02 Apr '14

Change in v2: - Add .get_regmap() support for struct snd_soc_codec_driver. Xiubo Li (11): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: 88pm860x: Remove the set_cache_io() entirely from ASoC probe. ASoc: cq93vc: Remove the set_cache_io() entirely from ASoC probe. ASoc: mc13783: Remove the set_cache_io() entirely from ASoC probe. ASoc: si476x: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5102: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5110: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8350: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8400: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8994: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8997: Remove the set_cache_io() entirely from ASoC probe. include/sound/soc.h | 1 + sound/soc/codecs/88pm860x-codec.c | 12 ++++++++---- sound/soc/codecs/cq93vc.c | 10 ++++++++-- sound/soc/codecs/mc13783.c | 14 ++++++-------- sound/soc/codecs/si476x.c | 14 ++++++-------- sound/soc/codecs/wm5102.c | 12 ++++++++---- sound/soc/codecs/wm5110.c | 12 ++++++++---- sound/soc/codecs/wm8350.c | 10 ++++++++-- sound/soc/codecs/wm8400.c | 10 ++++++++-- sound/soc/codecs/wm8994.c | 10 ++++++++-- sound/soc/codecs/wm8997.c | 13 ++++++++----- sound/soc/soc-core.c | 28 ++++++++++++++++++---------- sound/soc/soc-io.c | 9 +++------ 13 files changed, 98 insertions(+), 57 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH 00/13] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 02 Apr '14

02 Apr '14

Hi, This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. - Lars Lars-Peter Clausen (13): ASoC: Add snd_soc_kcontrol_codec() helper function ASoC: Add snd_soc_kcontrol_platform() helper function ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap ASoC: Move IO functions to soc-io.c ASoC: Drop ASoC level caching from hw_write/hw_read ASoC: Remove IO register modifier callbacks ASoC: Add helper function to cast component back to CODEC ASoC: Track which components have been registered with snd_soc_register_component() ASoC: Let snd_soc_platform subclass snd_soc_component ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() include/sound/soc-dapm.h | 1 + include/sound/soc.h | 114 +++++++-- sound/soc/codecs/88pm860x-codec.c | 8 +- sound/soc/codecs/ab8500-codec.c | 12 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs42l51.c | 4 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/lm4857.c | 4 +- sound/soc/codecs/max9768.c | 4 +- sound/soc/codecs/max98088.c | 12 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/pcm1681.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta32x.c | 4 +- sound/soc/codecs/tas5086.c | 4 +- sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/tlv320dac33.c | 4 +- sound/soc/codecs/twl4030.c | 10 +- sound/soc/codecs/twl6040.c | 8 +- sound/soc/codecs/wl1273.c | 12 +- sound/soc/codecs/wm2000.c | 8 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 14 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 32 +-- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8983.c | 4 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm_adsp.c | 4 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 8 +- sound/soc/soc-cache.c | 2 - sound/soc/soc-core.c | 520 ++++++++++++++++---------------------- sound/soc/soc-dapm.c | 91 +------ sound/soc/soc-io.c | 220 +++++++++++++--- 55 files changed, 644 insertions(+), 592 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH v8] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 02 Apr '14

02 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Daniel or Clemens: could I get another ACK on this? Thank you both! Best regards, Mario Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 477 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 494 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..9fda92d --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,477 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

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Re: [alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 02 Apr '14

02 Apr '14

> -----Original Message----- > From: Lars-Peter Clausen [mailto:lars@metafoo.de] > Sent: Tuesday, April 01, 2014 12:48 AM > To: Arun Shamanna Lakshmi > Cc: lgirdwood(a)gmail.com; broonie(a)kernel.org; swarren(a)wwwdotorg.org; > perex(a)perex.cz; tiwai(a)suse.de; alsa- devel(a)alsa-project.org; > linux-kernel(a)vger.kernel.org; Songhee Baek > Subject: Re: [PATCH] ASoC: DAPM: Add support for multi register mux > > On 04/01/2014 08:21 AM, Arun Shamanna Lakshmi wrote: > > Modify soc_enum struct to handle pointers for reg and mask. Add dapm > > get and put APIs for multi register mux with one hot encoding. > > > > Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> > > Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> > > Looks in my opinion much better than the previous version :) Just a > few minor issues, comments inline > > > --- > > include/sound/soc-dapm.h | 10 ++++ > > include/sound/soc.h | 22 +++++-- > > sound/soc/soc-core.c | 12 ++-- > > sound/soc/soc-dapm.c | 143 > +++++++++++++++++++++++++++++++++++++++++----- > > 4 files changed, 162 insertions(+), 25 deletions(-) > > > > diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h > index > > ef78f56..983b0ab 100644 > > --- a/include/sound/soc-dapm.h > > +++ b/include/sound/soc-dapm.h > > @@ -305,6 +305,12 @@ struct device; > > .get = snd_soc_dapm_get_enum_double, \ > > .put = snd_soc_dapm_put_enum_double, \ > > .private_value = (unsigned long)&xenum } > > +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ > > maybe just call it ENUM_ONEHOT, since it doesn't actually have to be > more than one register. > > [...] > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index > > cd52d52..aba0094 100644 > > --- a/sound/soc/soc-core.c > > +++ b/sound/soc/soc-core.c > > @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct > snd_kcontrol *kcontrol, > > unsigned int val, item; > > unsigned int reg_val; > > > > - reg_val = snd_soc_read(codec, e->reg); > > - val = (reg_val >> e->shift_l) & e->mask; > > + reg_val = snd_soc_read(codec, e->reg[0]); > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[0] = item; > > if (e->shift_l != e->shift_r) { > > - val = (reg_val >> e->shift_l) & e->mask; > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[1] = item; > > } > > @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct > snd_kcontrol *kcontrol, > > if (item[0] >= e->items) > > return -EINVAL; > > val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; > > - mask = e->mask << e->shift_l; > > + mask = e->mask[0] << e->shift_l; > > if (e->shift_l != e->shift_r) { > > if (item[1] >= e->items) > > return -EINVAL; > > val |= snd_soc_enum_item_to_val(e, item[1]) << e- shift_r; > > - mask |= e->mask << e->shift_r; > > + mask |= e->mask[0] << e->shift_r; > > } > > > > - return snd_soc_update_bits_locked(codec, e->reg, mask, val); > > + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); > > } > > EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); > > > > diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index > > c8a780d..4d2b35c 100644 > > --- a/sound/soc/soc-dapm.c > > +++ b/sound/soc/soc-dapm.c > > @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct > snd_soc_dapm_context *dapm, > > unsigned int val, item; > > int i; > > > > - if (e->reg != SND_SOC_NOPM) { > > - soc_widget_read(dest, e->reg, &val); > > - val = (val >> e->shift_l) & e->mask; > > + if (e->reg[0] != SND_SOC_NOPM) { > > + soc_widget_read(dest, e->reg[0], &val); > > + val = (val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > This probably should handle the new enum type as well. You'll probably > need some kind of flag in the struct to distinguish between the two > enum types. Any suggestion on the flag name ? > > > } else { > > /* since a virtual mux has no backing registers to > [...] > > /** > > + * snd_soc_dapm_get_enum_wide - dapm semi enumerated multiple > > + registers > > What's a semi-enumerated register? > > > + * mixer get callback > > + * @kcontrol: mixer control > > + * @ucontrol: control element information > > + * > > + * Callback to get the value of a dapm semi enumerated multiple > > +register mixer > > + * control. > > + * > > + * semi enumerated multiple registers mixer: > > + * the mixer has multiple registers to set the enumerated items. > > +The enumerated > > + * items are referred as values. > > + * Can be used for handling bit field coded enumeration for example. > > + * > > + * Returns 0 for success. > > + */ > > +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int reg_val, val, bit_pos = 0, reg_idx; > > + > > + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { > > + reg_val = snd_soc_read(codec, e->reg[reg_idx]); > > + val = reg_val & e->mask[reg_idx]; > > + if (val != 0) { > > + bit_pos = ffs(val) + (e->reg_width * reg_idx); > > Should be __ffs. __ffs returns the bits zero-indexed and ffs one-indexed. > That will work better for cases where there is not additional value > table necessary, since it means bit 1 maps to value 0. > > > + break; > > + } > > + } > > + > > + ucontrol->value.enumerated.item[0] = > > + snd_soc_enum_val_to_item(e, bit_pos); > > + > > + return 0; > > +} > [...] > > +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct snd_soc_card *card = codec->card; > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int *item = ucontrol->value.enumerated.item; > > + unsigned int change = 0, reg_idx = 0, value, bit_pos; > > + struct snd_soc_dapm_update update; > > + int ret = 0, reg_val = 0, i; > > + > > + if (item[0] >= e->items) > > + return -EINVAL; > > + > > + value = snd_soc_enum_item_to_val(e, item[0]); > > + > > + if (value) { > > + /* get the register index and value to set */ > > + reg_idx = (value - 1) / e->reg_width; > > + bit_pos = (value - 1) % e->reg_width; > > Changing the ffs to __ffs also means you can drop the ' - 1' here. > > Also e->reg_width should be (codec->val_bytes * 8) and reg_width field > should be dropped from the enum struct. > > > + reg_val = BIT(bit_pos); > > + } > > + > > + for (i = 0; i < e->num_regs; i++) { > > + if (i == reg_idx) { > > + change = snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], > reg_val); > > + > > + } else { > > + /* accumulate the change to update the DAPM > path > > + when none is selected */ > > + change += snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], 0); > > change |= > > > + > > + /* clear the register when not selected */ > > + snd_soc_write(codec, e->reg[i], 0); > > I think this should happen as part of the DAPM update sequence like > you had earlier. Some special care should probably be take to make > sure that you de-select the previous mux input before selecting the > new one if the new one is in a different register than the previous one. I am not sure I follow this part. We are clearing the 'not selected' registers before we set the one we want. Do you want us to loop the logic of soc_dapm_mux_update_power for each register ? or do you want to change the dapm_update structure so that it takes all the regs, masks, and values together ? > > > + } > > + } > > + > > + mutex_lock_nested(&card->dapm_mutex, > SND_SOC_DAPM_CLASS_RUNTIME); > > + > [...]

3 8

[alsa-devel] [PATCH] ASoC: core: Only kmemdup binary control buffer if masking
by Charles Keepax 02 Apr '14

02 Apr '14

When writing a binary control we may apply a mask to the first register, as this requires modifying the data the buffer is duplicated, currently this is done for all binary control writes. As most binary controls don't use the mask facility and thus can freely use the original buffer, avoid the kmemdup for these cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 86 ++++++++++++++++++++++++------------------------- 1 files changed, 42 insertions(+), 44 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..275bd71 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3229,67 +3229,65 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, len = params->num_regs * codec->val_bytes; - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; + if (!params->mask) + return regmap_raw_write(codec->control_data, params->base, + ucontrol->value.bytes.data, len); /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a * copy. */ - if (params->mask) { - ret = regmap_read(codec->control_data, params->base, &val); - if (ret != 0) - goto out; + data = kmemdup(ucontrol->value.bytes.data, len, + GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; - val &= params->mask; + ret = regmap_read(codec->control_data, params->base, &val); + if (ret != 0) + goto out; - switch (codec->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + val &= params->mask; - ((u16 *)data)[0] &= mask; + switch (codec->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u16 *)data)[0] &= mask; - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) + goto out; - ((u32 *)data)[0] &= mask; + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u32 *)data)[0] &= mask; - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) goto out; - } + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; } - ret = regmap_raw_write(codec->control_data, params->base, - data, len); + ret = regmap_raw_write(codec->control_data, params->base, data, len); out: kfree(data); -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: core: Fail codec probe if we fail to add controls
by Nariman Poushin 02 Apr '14

02 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35c0e71..7f9ae6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1127,9 +1127,17 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); + if (driver->dapm_widgets) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + driver->dapm_widgets, + driver->num_dapm_widgets); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create new controls %d\n", ret); + goto err_probe; + } + } /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &codec->component.dai_list, list) { -- 1.8.5.3

2 2

Re: [alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by Clemens Ladisch 02 Apr '14

02 Apr '14

Andoru wrote: > The last time you replied you said that it's a hardware error. If it > is that way then how come the card otherwise works flawlessly in > Windows, and this doesn't happen on Debian when watching videos or > with certain programs? It's still a hardware error, but, apparently, it happens only when the Linux driver accesses it in a certain way. Which programs work? Which do not? Regards, Clemens

3 3

[alsa-devel] [PATCH bisect 0/2] ASoC: fsl_sai: Overwrite trigger()
by Nicolin Chen 02 Apr '14

02 Apr '14

This series of patches are bisected from the preivous version so as to apply the PATCH-1 onto asoc-v3.15-4. * The patches are generated by using '-U2' because the default '-U3' would conflict the baseline without fsl_sai_isr patches. Nicolin Chen (2): ASoC: fsl_sai: Fix buggy configurations in trigger() ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) -- 1.8.4

2 9

[alsa-devel] pulseaudio and rcar-h2 sound
by Ben Dooks 01 Apr '14

01 Apr '14

I am currently investigating an issue with the rcar-h2 audio driver and a hang when running pulse-audio. From what I can see the issue seems to be an interaction with the driver and the regmap code. The current issue is that whatever is happening causes CPU0 to stall, thus nothing seems to be able to continue. The following errors are detected by the rcu code: INFO: rcu_sched self-detected stall on CPU 0: (1 GPs behind) idle=415/2/0 softirq=3076/3077 (t=10500 jiffies g=520 c=519 q=51) I have been through the regmap code and the locking looks ok. I also enabled as much of the locking checks as I could and it does not produce any useful output as to why it would be deadlocking. I can run the system fine if I replace the rsnd_read, rsnd_write and rsnd_bset code with versions which do not directly use regmap. Does anyone have any experience with this, or any further ideas on how to continue debugging? -- Ben Dooks http://www.codethink.co.uk/ Senior Engineer Codethink - Providing Genius

1 0

Re: [alsa-devel] twl4030 latency update
by Peter Ujfalusi 01 Apr '14

01 Apr '14

Hi Leonardo, On 04/01/2014 12:41 PM, Leonardo Gabrielli wrote: > Dear Peter, > I actually managed to nearly halve the latency with McBSP2 and jackd with a > little trick: requesting 4 channels audio. Of course two will be zero, but the > FIFO will grow up more quick. Yes, this is expected. The FIFO is lost (or word) based. If you have mono audio, it is 1280 samples long, in stereo it is 640 samples with 4 channel it can hold up to 320 samples. > Outcome: > 4 channels, 44100, 64 input --> jack --> output takes 10.1ms latency, i.e.: > - 2 periods for input: 128 L/R frames > - 1280 FIFO words / 4 channels = 320 frames > Without the trick the latency was 17.4ms :) What you can also try is to reduce the max_rx_thres to let's say 10 samples (20 or 40 depending on the number of channels you are using). Keep the max_tx_thres as you have been using. > > CPU load as increased slightly in jackd, but of course any other jack client > will stay the same assumed that the 2 fake extra channels are unused. > > In this experiment sample size = 16-bit. If I try to default to 32bit nothing > changes (so I assume the FIFO words are 32-bit wide and can contain either > 16-bit zeropadded samples or 32, which I remember from the AM37xx tech guide > but I'm too lazy to check again ;) ) Yes it is like that. > Glitches may happen when some regular user tasks (scp, curl, ...) request some > cpu. I may try the new 3.14 kernel with SCHED_DEADLINE to see if jackd and > clients can really be not preempted by user tasks... > > Best > > Leonardo > > > > On 26/03/2014 13:51, Peter Ujfalusi wrote: >> On 03/26/2014 11:45 AM, Leonardo Gabrielli wrote: >>> On 26/03/2014 09:26, Peter Ujfalusi wrote: >>>> The McBSP2 FIFO will be always there. There's nothing can be done on that. >>>> The >>>> size on McBSP2 is 1280 words -> 640 stereo samples, ie ~29ms with 22050, >>>> 14.5ms with 44100. >>>> >>>> If you are staying in element mode this means that it is granted that the >>>> sample at the DMA pointer will out on the i2s line about the mentioned times. >>>> This is the delay caused by the FIFO itself. From where the rest is coming >>>> I'm >>>> not really sure. >>> BTW: I forgot to mention: the latency listed in my previous email is >>> input+output (i.e. I record pulses from the beagleboard input jack and the >>> delayed version to the beagleboard output jack). The twl4030 analog and >>> digital loopback features have been of course disabled, in order to get the >>> total latency due from A/D to D/A. >> This means that the McBSP latency in worst case is 1280 + selected rx >> threshold in words (so /2 in case of stereo.) If you lower the rx threshold >> you decrease the latency on the capture side. On the playback side there's >> nothing can be done. >> >>> So just to get confirm I understood the McBSP mechanism well: even though I >>> can transfer to/from DMA samples in bursts of <threshold> length, each sample >>> will always "travel along" the whole FIFO buffer length, (as if in a delay >>> line) and thus they will always have 640samples delay? >> On the playback side this is pretty much true. On capture side the threshold >> means that DMA will read from FIFO when threshold amount is available in it. >> >>> Would it be possible to workaround this, e.g. by putting 4-channel audio >>> frames instead of stereo frames in the FIFO (with 2 channels unused), in order >>> to fill up the FIFO more quickly and have less latency? Or is it pure craze? >> >From the FIFO McBSP takes data word by word. If you play stereo, you need to >> have stereo data in the FIFO. You can not skip two words with McBSP. >> >> The thing I tried for playback and did not worked AFAIR: >> In general the idea was to configure DMA to send threshold/channel to every >> request while configuring the McBSP threshold register to be 1280 - threshold. >> In case of threshold 80 (40 stereo samples) it would play out: >> transfer 40 samples to FIFO per DMA request >> assert the DMA request when we have space for 1260 (630 samples). The number >> is just a guess, keeping 10 samples in FIFO sounds safe enough >> This would keep the FIFO fill between 10 and 50 samples. >> But this does not work, I think McBSP is counting the received words also and >> deasserts the DMA request based on this count and not the FIFO level. >> >> Another thing which would be even more complicated is to play with the McBSP >> threshold runtime. With the same 40 sample: >> DMA is to transfer 40 samples per DMA requests. >> start >> 1. McBSP threshold to 80 >> 2. in dma interrupt callback McBSP threshold to 1260 >> 3. in McBSP warning interrupt (that we will be reaching the threshold soon) >> back to 80 >> 4. goto 2 >> >> If we could do the step between 3 and 4 within one sample time this might work >> but as soon as you are late the thing will fail. >> >> I know this is working in realtime systems like in DSPs and non linux >> systems... >> > -- Péter

1 0

[alsa-devel] [PATCH] ASoC: core: Fail probe if we fail to add dai widgets
by Nariman Poushin 01 Apr '14

01 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b322cf2..5cf7107 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1132,8 +1132,15 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) - snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + list_for_each_entry(dai, &codec->component.dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; + } + } codec->dapm.idle_bias_off = driver->idle_bias_off; -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: alc5632: fix uninit var in alc5632_probe()
by Stephen Warren 01 Apr '14

01 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> alc5632_probe() returns ret, yet it is not initialized or set anywhere. This ends up causing the function to appear to fail, and audio not to work on the Toshiba AC100, with my compiler at least. This function used to set ret in all cases, but recent cleanup removed that. Cc: Xiubo Li <Li.Xiubo(a)freescale.com> Fixes: 5d6be5aa6bec ("ASoC: codec: Simplify ASoC probe code.") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6306ef..3ee8d4e41a99 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.8.1.5

3 2

[alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 01 Apr '14

01 Apr '14

Modify soc_enum struct to handle pointers for reg and mask. Add dapm get and put APIs for multi register mux with one hot encoding. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 10 ++++ include/sound/soc.h | 22 +++++-- sound/soc/soc-core.c | 12 ++-- sound/soc/soc-dapm.c | 143 +++++++++++++++++++++++++++++++++++++++++----- 4 files changed, 162 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..983b0ab 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_wide, \ + .put = snd_soc_dapm_put_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -378,6 +384,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..5c274c4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,23 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1 } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1 } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, .reg_width = 32 } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +298,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_WIDE_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -1098,13 +1106,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + unsigned int reg_width; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..aba0094 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..4d2b35c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, unsigned int val, item; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; + if (e->reg[0] != SND_SOC_NOPM) { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); } else { /* since a virtual mux has no backing registers to @@ -2903,15 +2903,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,25 +2945,25 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; + update.reg = e->reg[0]; update.mask = mask; update.val = val; card->update = &update; @@ -2984,6 +2984,123 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_wide - dapm semi enumerated multiple registers + * mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = 0, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = ffs(val) + (e->reg_width * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_wide); + +/** + * snd_soc_dapm_put_enum_wide - dapm semi enumerated multiple registers + * mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + struct snd_soc_dapm_update update; + int ret = 0, reg_val = 0, i; + + if (item[0] >= e->items) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value) { + /* get the register index and value to set */ + reg_idx = (value - 1) / e->reg_width; + bit_pos = (value - 1) % e->reg_width; + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change = snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change += snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + snd_soc_write(codec, e->reg[i], 0); + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.reg = e->reg[reg_idx]; + update.mask = e->mask[reg_idx]; + update.val = reg_val; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_wide); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 01 Apr '14

01 Apr '14

> -----Original Message----- > From: Songhee Baek > Sent: Friday, March 28, 2014 11:10 AM > To: 'Lars-Peter Clausen' > Cc: Arun Shamanna Lakshmi; 'lgirdwood(a)gmail.com'; 'broonie(a)kernel.org'; > 'swarren(a)wwwdotorg.org'; 'alsa-devel(a)alsa-project.org'; 'tiwai(a)suse.de'; > 'linux-kernel(a)vger.kernel.org' > Subject: RE: [alsa-devel] [PATCH] ASoC: Add support for multi register mux > > > > > On 03/26/2014 11:41 PM, Songhee Baek wrote: > > > >> On 03/26/2014 01:02 AM, Arun Shamanna Lakshmi wrote: > > > >> > > > >> The way you describe this it seems to me that a value array for > > > >> this kind of mux would look like. > > > >> > > > >> 0x00000000, 0x00000000, 0x00000001 0x00000000, 0x00000000, > > > >> 0x00000002 0x00000000, 0x00000000, 0x00000003 0x00000000, > > > >> 0x00000000, 0x00000004 0x00000000, 0x00000000, 0x00000008 ... > > > >> > > > >> That seems to be extremely tedious. If the MUX uses a one hot > > > >> encoding how about storing the index of the bit in the values > > > >> array and use (1 << value) when writing the value to the register? > > > > > > > > If we store the index of the bit, the value will be duplicated for > > > > each > > > registers inputs since register has 0 to 31bits to shift, then we > > > need to decode the index to interpret value for which registers to > > > set. If we need to interpret the decoded value of index, it is > > > better to have custom put/get function in our driver, isn't it? > > > > > > > > > > I'm not sure I understand. If you use (val / 32) to pick the > > > register and (val % > > > 32) to pick the bit in the register this should work just fine. > > > Maybe I'm missing something. Do you have a real world code example > > > of of the this type of enum is used? > > > > > > > I can use val/32 and val%32 for this multi register mux. > With this logic, put function would be easy however get function becomes tedious due to looping on each bit per register for 3 of them in our case. Rather, it would be easy to identify a unique MUX_OFFSET to distinguish between the 3 registers as shown in the code below. These get/put functions are updated from previous mail, now it works for multi register mux, please review these function whether I can add in current put/get function. #define MULTI_MUX_INPUT_OFFSET(n) (5 * n) int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, reg_idx; if (e->reg[0] != SND_SOC_NOPM) { for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { reg_val = snd_soc_read(codec, e->reg[reg_idx]); val = (reg_val >> e->shift_l) & e->mask[reg_idx]; if (val) { val += MULTI_MUX_INPUT_OFFSET(reg_idx); break; } } } else { reg_val = dapm_kcontrol_get_value(kcontrol); val = (reg_val >> e->shift_l) & e->mask[0]; } ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } return 0; } int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int change, i, value, val, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { value = val; goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = val - MULTI_MUX_INPUT_OFFSET(i); /* checking reg_val is power of 2 : one-hot code */ /* if reg_val after subtract MULTI_MUX_INPUT_OFFSET is not power of 2, reg[i] should be zero */ if (reg_val & (reg_val - 1)) { /* clear the current input register */ snd_soc_write(codec, e->reg[i], 0); } else { /* reg_val is power of 2, store updated info */ value = reg_val; mask = e->mask[i]; update_idx = i; } } update_reg: mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg[update_idx] != SND_SOC_NOPM) change = snd_soc_test_bits(codec, e->reg[update_idx], mask, value); else change = dapm_kcontrol_set_value(kcontrol, value); if (change) { if (e->reg[update_idx] != SND_SOC_NOPM) { update.kcontrol = kcontrol; update.reg = e->reg[update_idx]; update.mask= mask; update.val = value; card->update = &update; } ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); return change; } > > > >>> - int reg; > > > >>> + int reg[SOC_ENUM_MAX_REGS]; > > > >>> unsigned char shift_l; > > > >>> unsigned char shift_r; > > > >>> unsigned int items; > > > >>> - unsigned int mask; > > > >>> + unsigned int mask[SOC_ENUM_MAX_REGS]; > > > >> > > > >> If you make mask and reg pointers instead of arrays this should > > > >> be much more flexible and not be limited to 3 registers. > We will make reg* and mask* instead of arrays and since we use the same structure, the plan is to share the get and put function code. Thanks. Songhee.

4 6

Re: [alsa-devel] [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams
by Nicolin Chen 01 Apr '14

01 Apr '14

On Tue, Apr 01, 2014 at 11:48:16AM +0800, Wang Dongsheng-B40534 wrote: > > > > -----Original Message----- > > From: Nicolin Chen [mailto:Guangyu.Chen@freescale.com] > > Sent: Tuesday, April 01, 2014 11:14 AM > > To: Wang Dongsheng-B40534 > > Cc: broonie(a)kernel.org; alsa-devel(a)alsa-project.org; Xiubo Li-B47053; linuxppc- > > dev(a)lists.ozlabs.org; linux-kernel(a)vger.kernel.org; timur(a)tabi.org > > Subject: Re: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx and Rx streams > > > > On Tue, Apr 01, 2014 at 11:25:02AM +0800, Wang Dongsheng-B40534 wrote: > > > > Subject: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx > > > > and Rx streams > > > > > > > > We only enable one side interrupt for each stream since over/underrun > > > > on the opposite stream would be resulted from what we previously did, > > > > enabling TERE but remaining FRDE disabled, even though the xrun on the > > > > opposite direction will not break the current stream. > > > > > > > > Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> > > > > Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> > > > > --- > > > > sound/soc/fsl/fsl_sai.c | 8 ++++++-- > > > > sound/soc/fsl/fsl_sai.h | 1 + > > > > 2 files changed, 7 insertions(+), 2 deletions(-) > > > > > > > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > > > > index bdfd497..d64c33f 100644 > > > > --- a/sound/soc/fsl/fsl_sai.c > > > > +++ b/sound/soc/fsl/fsl_sai.c > > > > @@ -397,4 +397,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); > > > > break; > > > > @@ -404,4 +406,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, 0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, 0); > > > > > > > > if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { > > > > @@ -464,6 +468,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) > > > > struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); > > > > > > > > - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); > > > > - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); > > > > > > Why are you remove this macro? Don't use magic number. > > > > It's pretty clear that the so-called magic number is to clear the settings > > in the registers for driver init as what this driver did at the first place > > -- no offense but I don't think you would ask this if you check the git-log > > of the driver. > > > ~FSL_SAI_MASK is better than 0x0. And you also replace 0xffffffff. I would later send a patch to reset SAI for a true init instead of these lines but not within this patch as it's focusing on the interrupts enabling. So please don't grasp the mask here. Just let me continue. Thank you, Nicolin Chen

1 0

Re: [alsa-devel] [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams
by Nicolin Chen 01 Apr '14

01 Apr '14

On Tue, Apr 01, 2014 at 11:25:02AM +0800, Wang Dongsheng-B40534 wrote: > > Subject: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx > > and Rx streams > > > > We only enable one side interrupt for each stream since over/underrun > > on the opposite stream would be resulted from what we previously did, > > enabling TERE but remaining FRDE disabled, even though the xrun on the > > opposite direction will not break the current stream. > > > > Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> > > Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> > > --- > > sound/soc/fsl/fsl_sai.c | 8 ++++++-- > > sound/soc/fsl/fsl_sai.h | 1 + > > 2 files changed, 7 insertions(+), 2 deletions(-) > > > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > > index bdfd497..d64c33f 100644 > > --- a/sound/soc/fsl/fsl_sai.c > > +++ b/sound/soc/fsl/fsl_sai.c > > @@ -397,4 +397,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > *substream, int cmd, > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); > > break; > > @@ -404,4 +406,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > *substream, int cmd, > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > FSL_SAI_CSR_FRDE, 0); > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > + FSL_SAI_CSR_xIE_MASK, 0); > > > > if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { > > @@ -464,6 +468,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) > > struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); > > > > - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); > > - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); > > + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); > > + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); > > Why are you remove this macro? Don't use magic number. It's pretty clear that the so-called magic number is to clear the settings in the registers for driver init as what this driver did at the first place -- no offense but I don't think you would ask this if you check the git-log of the driver. Thank you, Nicolin Chen

1 0

[alsa-devel] [PATCH] ALSA: usb-audio - Capture and duplex support for Digidesign Mbox 1 sound card.
by Damien Zammit 01 Apr '14

01 Apr '14

This patch completes the support for Digidesign Mbox 1 sound card by adding capture. The issue with this card was that it has two endpoints on the same interface. This patch creates a dual endpoint quirk. The quirk interface needs a second audioformat struct for this to work which I called ".data2". Signed-off-by: Damien Zammit <damien(a)zamaudio.com>

4 13

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix buggy configurations in trigger()
by Nicolin Chen 01 Apr '14

01 Apr '14

The current trigger() has two crucial problems: 1) The DMA request enabling operations (FSL_SAI_CSR_FRDE) for Tx and Rx are now totally exclusive: It would fail to run simultaneous Tx-Rx cases. 2) The TERE disabling operation depends on an incorrect condition -- active reference count that only gets increased in snd_pcm_open() and decreased in snd_pcm_close(): The TERE would never get cleared. So this patch overwrites the trigger function by following these rules: A) We continue to support tx-async-while-rx-sync-to-tx case alone, which's originally limited by this fsl_sai driver, but we make the code easy to modify for the further support of the opposite case. B) We enable both TE and RE for PLAYBACK stream or CAPTURE stream but only enabling the DMA request bit (FSL_SAI_CSR_FRDE) of the current direction due to the requirement of SAI -- For tx-async-while-rx-sync-to-tx case, the receiver is enabled only when both the transmitter and receiver are enabled. C) We only enable one side interrupt for each stream since over/underrun on the opposite stream would be resulted from what we've done in rule B: enabling TERE but remaining FRDE disabled, even though the xrun on the opposite direction will not break the current stream. Tested cases: a) aplay test.wav -d5 b) arecord -r44100 -c2 -fS16_LE test.wav -d5 c) arecord -r44100 -c2 -fS16_LE -d5 | aplay d) (aplay test2.wav &); sleep 1; arecord -r44100 -c2 -fS16_LE test.wav -d1 e) (arecord -r44100 -c2 -fS16_LE test.wav -d5 &); sleep 1; aplay test.wav -d1 Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index f088545..d64c33f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -365,6 +365,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 tcsr, rcsr; /* @@ -379,14 +380,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - tcsr |= FSL_SAI_CSR_FRDE; - rcsr &= ~FSL_SAI_CSR_FRDE; - } else { - rcsr |= FSL_SAI_CSR_FRDE; - tcsr &= ~FSL_SAI_CSR_FRDE; - } - /* * It is recommended that the transmitter is the last enabled * and the first disabled. @@ -395,22 +388,32 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - tcsr |= FSL_SAI_CSR_TERE; - rcsr |= FSL_SAI_CSR_TERE; + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { - tcsr &= ~FSL_SAI_CSR_TERE; - rcsr &= ~FSL_SAI_CSR_TERE; + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, 0); + + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, 0); } - - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -464,8 +467,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index a264185..be26d46 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -35,6 +35,16 @@ #define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ #define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ +#define FSL_SAI_xCSR(tx) (tx ? FSL_SAI_TCSR : FSL_SAI_RCSR) +#define FSL_SAI_xCR1(tx) (tx ? FSL_SAI_TCR1 : FSL_SAI_RCR1) +#define FSL_SAI_xCR2(tx) (tx ? FSL_SAI_TCR2 : FSL_SAI_RCR2) +#define FSL_SAI_xCR3(tx) (tx ? FSL_SAI_TCR3 : FSL_SAI_RCR3) +#define FSL_SAI_xCR4(tx) (tx ? FSL_SAI_TCR4 : FSL_SAI_RCR4) +#define FSL_SAI_xCR5(tx) (tx ? FSL_SAI_TCR5 : FSL_SAI_RCR5) +#define FSL_SAI_xDR(tx) (tx ? FSL_SAI_TDR : FSL_SAI_RDR) +#define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) +#define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) + /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) @@ -48,6 +58,7 @@ #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRF BIT(16) #define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_xIE_MASK (0x1f << FSL_SAI_CSR_xIE_SHIFT) #define FSL_SAI_CSR_WSIE BIT(12) #define FSL_SAI_CSR_SEIE BIT(11) #define FSL_SAI_CSR_FEIE BIT(10) -- 1.8.4

3 2

[alsa-devel] cloning snd_seq_t (or creating one from client id)
by Tobias Schlemmer 31 Mar '14

31 Mar '14

Hi, Recently, I came upon the following problem with the ALSA sequencer interface: I'm surprised that the upcoming version of RtMidi uses one ALSA client per MIDI port. As an end user I'd expect to see only one ALSA client per program instance (e.g. using aconnect -i - o -l). Asking for the reason of that change I got the answer that it has something to do with broken multithread support. Digging deeper into this problem I found out that sequencer handles (snd_seq_t) are not thread safe. So my question is: Could you provide some way to implement multithreaded ALSA clients? My idea is to provide some function int snd_seq_clone(snd_seq_t * old, snd_seq_t ** new) /* clone a sequencer */ and/or int snd_seq_create_from_client(int clientid, snd_seq_t ** new) /* create a sequencer from a given client id */ and/or int snd_seq_split_port(snd_seq_t * old, snd_seq_t ** new, int port) /* create a sequencer that steels a port from old */ For the semantics I think cloned sequencers should act as if they don't know each other. Each one has its own set of ports. While the generation of port numbers still uses the same counter. I don't expect creating a port to be that time critical that it can't use locks. If it helps to ease the implementation one could use a master sequencer with some slaves. If the master sequencer is deleted the slaves are automatically freed and must not be used anymore. And if a subsequencer tries to access a port of another subsequncer, ALSA could just return with an error. Regards, Tobias P.S.: please, CC to me in answers.

3 3

[alsa-devel] [PATCH] ASoC: rt5640: Change the setting method of idle_bias_off
by Oder Chiou 31 Mar '14

31 Mar '14

The patch moves the idle_bias_off setting to struct "soc_codec_dev_rt5640". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b088f14..6b8778f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1895,7 +1895,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); @@ -2004,6 +2003,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .suspend = rt5640_suspend, .resume = rt5640_resume, .set_bias_level = rt5640_set_bias_level, + .idle_bias_off = true, .controls = rt5640_snd_controls, .num_controls = ARRAY_SIZE(rt5640_snd_controls), .dapm_widgets = rt5640_dapm_widgets, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt5640: add an of_match table
by Stephen Warren 31 Mar '14

31 Mar '14

From: Stephen Warren <swarren(a)nvidia.com> Add a device tree match table. This serves to make the driver's support of device tree more explicit. Perhaps the fallback for DT matching to using the i2c_device_id table will go away one day, since it fails in face of devices from different vendors with the same name. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b5014a2024e2..b6a02c16f100 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2018,6 +2018,14 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5640", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5640_of_match); +#endif + #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, @@ -2166,6 +2174,7 @@ static struct i2c_driver rt5640_i2c_driver = { .name = "rt5640", .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), + .of_match_table = of_match_ptr(rt5640_of_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, -- 1.8.1.5

3 3

[alsa-devel] [PATCH] ASoC: rt5640: Use the platform data for DMIC settings
by Oder Chiou 31 Mar '14

31 Mar '14

The patch uses the platform data for DMIC settings. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5640.h | 4 +++ sound/soc/codecs/rt5640.c | 77 ++++++++++++++--------------------------------- 2 files changed, 27 insertions(+), 54 deletions(-) diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 27cc75e..59d26dd 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -16,6 +16,10 @@ struct rt5640_platform_data { bool in1_diff; bool in2_diff; + bool dmic_en; + bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ + bool dmic2_data_pin; /* 0 = IN1N; 1 = GPIO4 */ + int ldo1_en; /* GPIO for LDO1_EN */ }; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index bcd6513..cf041e3 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -872,54 +872,6 @@ static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP, static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK | RT5640_GP3_PIN_MASK, - RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP3_PIN_DMIC1_SDA); - snd_soc_update_bits(codec, RT5640_DMIC, - RT5640_DMIC_1L_LH_MASK | RT5640_DMIC_1R_LH_MASK | - RT5640_DMIC_1_DP_MASK, - RT5640_DMIC_1L_LH_FALLING | RT5640_DMIC_1R_LH_RISING | - RT5640_DMIC_1_DP_IN1P); - break; - - default: - return 0; - } - - return 0; -} - -static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK | RT5640_GP4_PIN_MASK, - RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP4_PIN_DMIC2_SDA); - snd_soc_update_bits(codec, RT5640_DMIC, - RT5640_DMIC_2L_LH_MASK | RT5640_DMIC_2R_LH_MASK | - RT5640_DMIC_2_DP_MASK, - RT5640_DMIC_2L_LH_FALLING | RT5640_DMIC_2R_LH_RISING | - RT5640_DMIC_2_DP_IN1N); - break; - - default: - return 0; - } - - return 0; -} - static void hp_amp_power_on(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1054,12 +1006,10 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, - RT5640_DMIC_1_EN_SFT, 0, rt5640_set_dmic1_event, - SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, - RT5640_DMIC_2_EN_SFT, 0, rt5640_set_dmic2_event, - SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, RT5640_DMIC_1_EN_SFT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, RT5640_DMIC_2_EN_SFT, 0, + NULL, 0), /* Boost */ SND_SOC_DAPM_PGA("BST1", RT5640_PWR_ANLG2, RT5640_PWR_BST1_BIT, 0, NULL, 0), @@ -2179,6 +2129,25 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.dmic_en) { + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (rt5640->pdata.dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (rt5640->pdata.dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + } + rt5640->hp_mute = 1; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- 1.8.1.1.439.g50a6b54

3 7

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 31 Mar '14

31 Mar '14

The following changes since commit e090d5b6ad20056ec0ef58727e3ae95fd82be090: Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next (2014-03-23 14:00:59 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-4 for you to fetch changes up to c159a85013afbb8283f0c7272812952e04d5c3a1: Merge remote-tracking branches 'asoc/topic/tdm' and 'asoc/topic/tegra' into asoc-next (2014-03-25 21:22:15 +0000) ---------------------------------------------------------------- ASoC: Final updates for v3.15 merge window A few more updates from last week - use of the tdm_slot mapping from Xiubo plus a few smaller fixes and cleanups. ---------------------------------------------------------------- Alex Smith (1): MIPS: Fix possible build error with transparent hugepages enabled Andreas Herrmann (1): MIPS: Octeon: Fix warning in of_device_alloc on cn3xxx Arnaldo Carvalho de Melo (1): perf bench numa: Make no args mean 'run all tests' Ben Pfaff (1): openvswitch: Correctly report flow used times for first 5 minutes after boot. Ben Widawsky (1): drm/i915: Fix PSR programming Benedikt Spranger (1): net: cpsw: do not register cpts twice Bjorn Helgaas (1): Revert "[PATCH] Insert GART region into resource map" Bjørn Mork (1): net: cdc_ncm: fix control message ordering Boris BREZILLON (1): ARM: at91: fix network interface ordering for sama5d36 Charles Keepax (1): ALSA: compress: Pass through return value of open ops callback Chris Wilson (1): drm/i915: Disable stolen memory when DMAR is active Christian Riesch (2): net: davinci_emac: Replace devm_request_irq with request_irq net: davinci_emac: Fix rollback of emac_dev_open() Clemens Ladisch (1): ALSA: oxygen: Xonar DG(X): fix Stereo Upmixing regression Colin Ian King (1): MIPS: Octeon: Fix fall through on bar type OCTEON_DMA_BAR_TYPE_SMALL Daniel Kurtz (1): drm/exynos: Fix (more) freeing issues in exynos_drm_drv.c Daniel Vetter (2): drm/i915: Don't enable display error interrupts from the start drm: Fix use-after-free in the shadow-attache exit code Dave Airlie (3): Merge tag 'drm-intel-fixes-2014-03-17' of git://anongit.freedesktop.org/drm-intel into drm-fixes Merge tag 'drm-intel-fixes-2014-03-19' of git://anongit.freedesktop.org/drm-intel into drm-fixes Merge branch 'exynos-drm-fixes' of git://git.kernel.org/.../daeinki/drm-exynos into drm-fixes Dave Jones (2): brcmfmac: fix skb leak in brcmf_sdio_txpkt_prep_sg error path. block: free q->flush_rq in blk_init_allocated_queue error paths Dave Kleikamp (1): sparc64: don't treat 64-bit syscall return codes as 32-bit David Miller (1): sparc: serial: Clean up the locking for -rt David S. Miller (6): Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'master' of git://git.kernel.org/.../klassert/ipsec Merge branch 'cnic-net' Merge branch 'fixes' of git://git.kernel.org/.../jesse/openvswitch sparc64: Make sure %pil interrupts are enabled during hypervisor yield. Merge branch 'davinci_emac' David Stevens (2): vxlan: fix potential NULL dereference in arp_reduce() vxlan: fix nonfunctional neigh_reduce() Deng-Cheng Zhu (4): MIPS: APRP: Fix the linking of rtlx interrupt hook MIPS: APRP: Unregister rtlx interrupt hook at module exit MIPS: APRP: Choose the correct VPE loader by fixing the linking MIPS: math-emu: Fix prefx detection and COP1X function field definition Doug Wilson (1): sparc64:tsb.c:use array size macro rather than number Eric Dumazet (1): tcp: syncookies: do not use getnstimeofday() Erik Hugne (1): tipc: fix spinlock recursion bug for failed subscriptions Felix Fietkau (1): ath9k_hw: fix unreachable code in baseband hang detection code Fernando Luis Vázquez Cao (1): HID: hidraw: fix warning destroying hidraw device files after parent Frank Praznik (1): HID: sony: Fix work queue issues. Helge Deller (2): partly revert commit 8a10bc9: parisc/sti_console: prefer Linux fonts over built-in ROM fonts parisc: wire up sys_utimes Helmut Schaa (1): ath9k: Fix sequence number assignment for non-data frames Huacai Chen (1): MIPS: FPU: Fix conflict of register usage Hugh Dickins (2): mm: fix bad rss-counter if remap_file_pages raced migration mm: fix swapops.h:131 bug if remap_file_pages raced migration Ingo Molnar (1): Merge tag 'perf-urgent-for-mingo' of git://git.kernel.org/.../acme/linux into perf/urgent Jani Nikula (1): Revert "drm/i915: don't touch the VDD when disabling the panel" Jarno Rajahalme (2): openvswitch: Read tcp flags only then the tranport header is present. openvswitch: Fix race. Jean-Francois Moine (3): ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Add DT documentation for multi-DAI links ASoC: simple-card: Handle many DAI links Jens Axboe (1): mtip32xx: fix bad use of smp_processor_id() Jim Quinlan (1): MIPS: Make local_irq_disable macro safe for non-Mipsr2 Jiri Pirko (1): ovs: fix dp check in ovs_dp_reset_user_features Joe Perches (1): MAINTAINERS: Add linux.nics(a)intel.com to INTEL ETHERNET DRIVERS John David Anglin (1): parisc: Remove unused CONFIG_PARISC_TMPALIAS code John W. Linville (1): Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Kuninori Morimoto (3): ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: rcar: bugfix: it cares about the non-src case ASoC: rcar: subnode tidyup for renesas,rsnd.txt Lars Persson (1): MIPS: Fix syscall tracing interface Lars-Peter Clausen (1): ASoC: Remove name_prefix unset during DAI link init hack Li RongQing (1): netpoll: fix the skb check in pkt_is_ns Li Zefan (1): cgroup: fix a failure path in create_css() Linus Torvalds (18): Merge branch 'for-linus' of git://git.kernel.org/.../jikos/hid Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge branch 'for-linus' of git://git.kernel.dk/linux-block Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'for-3.14-fixes' of git://git.kernel.org/.../tj/cgroup Merge tag 'pci-v3.14-fixes-3' of git://git.kernel.org/.../helgaas/pci Merge tag 'sound-3.14' of git://git.kernel.org/.../tiwai/sound Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'trace-fixes-v3.14-rc7' of git://git.kernel.org/.../rostedt/linux-trace futex: revert back to the explicit waiter counting code Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge git://git.kernel.org/.../davem/net Merge git://git.kernel.org/.../davem/sparc Merge branch 'parisc-3.14' of git://git.kernel.org/.../deller/parisc-linux Linux 3.14-rc8 Lucas Stach (1): ASoC: tegra: move AC97 clock handling to the machine driver Manuel Lauss (1): MIPS: Alchemy: Fix unchecked kstrtoul return value Mark Brown (14): Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge tag 'asoc-v3.15-3' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge tag 'asoc-v3.15-3' into asoc-next Merge tag 'asoc-v3.15-rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/prefix' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branches 'asoc/topic/tdm' and 'asoc/topic/tegra' into asoc-next Markos Chandras (2): MIPS: bcm47xx: Include missing errno.h for ENXIO MIPS: asm: syscall: Fix copying system call arguments Michael Chan (3): cnic: Use proper ulp_ops for per device operations. cnic,bnx2i,bnx2fc: Fix inconsistent use of page size cnic: Update version to 2.5.20 and copyright year. Mike Snitzer (2): block: fix q->flush_rq NULL pointer crash on dm-mpath flush block: change flush sequence list addition back to front add Nicolas Dichtel (3): rtnetlink: fix fdb notification flags ipmr: fix mfc notification flags ip6mr: fix mfc notification flags Nikolay Aleksandrov (2): net: af_key: fix sleeping under rcu selinux: add gfp argument to security_xfrm_policy_alloc and fix callers Nishanth Menon (1): net: micrel : ks8851-ml: add vdd-supply support Patrick Palka (1): perf bench: Fix NULL pointer dereference in "perf bench all" Paul Bolle (2): MIPS: Replace CONFIG_MIPS64 and CONFIG_MIPS32_R2 isdn/capi: Make Middleware depend on CAPI2.0 Paul Burton (1): MIPS: mark O32+FP64 experimental for now Paul Mackerras (2): KVM: PPC: Book3S HV: Remove bogus duplicate code KVM: PPC: Book3S HV: Fix register usage when loading/saving VRSAVE Peter Senna Tschudin (2): ATHEROS-ALX: Use dma_set_mask_and_coherent and fix a bug ATHEROS-ATL1E: Convert iounmap to pci_iounmap Rafał Miłecki (1): MIPS: BCM47XX: Check all (32) GPIOs when looking for a pin Ralf Baechle (1): MIPS: Fix randconfig build error. Roger Luethi (1): via-rhine: Disable device in error path Sebastian Hesselbarth (1): net: phy: fix uninitalized ethtool_wolinfo in phy_suspend Shawn Guo (1): MAINTAINERS: update IMX kernel git tree Simon Wood (1): HID: hid-lg4ff: Support new version of G27 Stanislaw Gruszka (1): Revert "rt2x00: rt2800lib: Update BBP register initialization for RT53xx" Stefan Agner (1): clocksource: vf_pit_timer: use complement for sched_clock reading Vaibhav Nagarnaik (1): tracing: Fix array size mismatch in format string Ville Syrjälä (2): drm/i915: Add a workaround for HSW scanline counter weirdness drm/i915: Fix scanline counter fixup on BDW Viller Hsiao (2): MIPS: ftrace: Fix icache flush range error MIPS: ftrace: Tweak safe_load()/safe_store() macros Will Deacon (1): parisc: locks: remove redundant arch_*_relax operations Xiubo Li (4): ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask. ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support. lucien (1): ipv6: ip6_append_data_mtu do not handle the mtu of the second fragment properly stephen hemminger (1): netlink: fix setsockopt in mmap examples in documentation .../devicetree/bindings/net/micrel-ks8851.txt | 1 + .../devicetree/bindings/sound/renesas,rsnd.txt | 15 +- .../devicetree/bindings/sound/simple-card.txt | 42 +++- Documentation/networking/netlink_mmap.txt | 4 +- MAINTAINERS | 5 +- Makefile | 2 +- arch/arm/boot/dts/sama5d36.dtsi | 2 +- arch/arm/mach-shmobile/board-bockw.c | 16 +- arch/mips/Kconfig | 24 ++- arch/mips/alchemy/board-gpr.c | 4 +- arch/mips/alchemy/board-mtx1.c | 4 +- arch/mips/bcm47xx/board.c | 1 + arch/mips/bcm47xx/nvram.c | 2 +- arch/mips/cavium-octeon/octeon-irq.c | 22 ++- arch/mips/include/asm/asmmacro.h | 15 +- arch/mips/include/asm/fpu.h | 2 +- arch/mips/include/asm/ftrace.h | 20 +- arch/mips/include/asm/syscall.h | 10 +- arch/mips/include/uapi/asm/inst.h | 4 +- arch/mips/kernel/ftrace.c | 5 +- arch/mips/kernel/r4k_fpu.S | 16 +- arch/mips/kernel/rtlx-cmp.c | 3 + arch/mips/kernel/rtlx-mt.c | 3 + arch/mips/math-emu/cp1emu.c | 6 +- arch/mips/mti-malta/malta-amon.c | 2 +- arch/mips/mti-malta/malta-int.c | 4 +- arch/mips/pci/msi-octeon.c | 1 + arch/parisc/include/asm/page.h | 11 -- arch/parisc/include/asm/spinlock.h | 4 - arch/parisc/include/uapi/asm/unistd.h | 4 +- arch/parisc/kernel/cache.c | 64 ------ arch/parisc/kernel/syscall_table.S | 1 + arch/powerpc/kvm/book3s_hv_rmhandlers.S | 71 +------ arch/sparc/kernel/process_64.c | 4 +- arch/sparc/kernel/syscalls.S | 4 +- arch/sparc/mm/tsb.c | 2 +- arch/x86/kernel/aperture_64.c | 20 +- block/blk-core.c | 25 ++- block/blk-flush.c | 11 +- drivers/block/mtip32xx/mtip32xx.c | 2 +- drivers/clocksource/vf_pit_timer.c | 2 +- drivers/gpu/drm/drm_pci.c | 2 +- drivers/gpu/drm/exynos/exynos_drm_drv.c | 10 +- drivers/gpu/drm/i915/i915_gem_stolen.c | 7 + drivers/gpu/drm/i915/i915_irq.c | 71 ++++--- drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 12 +- drivers/hid/hid-lg4ff.c | 2 + drivers/hid/hid-sony.c | 27 ++- drivers/hid/hidraw.c | 4 +- drivers/isdn/capi/Kconfig | 18 +- drivers/net/ethernet/atheros/alx/main.c | 14 +- drivers/net/ethernet/atheros/atl1e/atl1e_main.c | 4 +- drivers/net/ethernet/broadcom/cnic.c | 111 +++++------ drivers/net/ethernet/broadcom/cnic.h | 2 +- drivers/net/ethernet/broadcom/cnic_defs.h | 2 +- drivers/net/ethernet/broadcom/cnic_if.h | 16 +- drivers/net/ethernet/micrel/ks8851.c | 30 ++- drivers/net/ethernet/ti/cpsw.c | 4 - drivers/net/ethernet/ti/davinci_cpdma.c | 4 +- drivers/net/ethernet/ti/davinci_emac.c | 53 +++-- drivers/net/ethernet/via/via-rhine.c | 8 +- drivers/net/phy/phy_device.c | 3 +- drivers/net/usb/cdc_ncm.c | 48 +++-- drivers/net/vxlan.c | 130 ++++++++++-- drivers/net/wireless/ath/ath9k/hw.c | 3 +- drivers/net/wireless/ath/ath9k/xmit.c | 4 +- drivers/net/wireless/brcm80211/brcmfmac/dhd_sdio.c | 4 +- drivers/net/wireless/rt2x00/rt2800lib.c | 6 +- drivers/scsi/bnx2fc/bnx2fc_io.c | 16 +- drivers/scsi/bnx2fc/bnx2fc_tgt.c | 38 ++-- drivers/scsi/bnx2i/bnx2i_hwi.c | 52 ++--- drivers/scsi/bnx2i/bnx2i_iscsi.c | 23 +-- drivers/tty/serial/sunhv.c | 22 +-- drivers/tty/serial/sunsab.c | 14 +- drivers/tty/serial/sunsu.c | 14 +- drivers/tty/serial/sunzilog.c | 14 +- include/linux/ftrace_event.h | 4 - include/linux/rmap.h | 3 +- include/linux/security.h | 10 +- include/linux/usb/cdc_ncm.h | 1 + include/net/tcp.h | 11 +- include/sound/soc-dai.h | 2 +- include/trace/ftrace.h | 7 +- kernel/cgroup.c | 11 +- kernel/futex.c | 53 ++++- kernel/trace/trace_events.c | 6 - kernel/trace/trace_export.c | 7 +- lib/fonts/Kconfig | 6 +- mm/fremap.c | 28 ++- mm/migrate.c | 32 +++ mm/rmap.c | 5 +- net/core/netpoll.c | 2 +- net/core/rtnetlink.c | 10 +- net/ipv4/ipmr.c | 13 +- net/ipv6/ip6_output.c | 14 +- net/ipv6/ip6mr.c | 13 +- net/key/af_key.c | 19 +- net/openvswitch/datapath.c | 9 +- net/openvswitch/flow.c | 3 +- net/tipc/subscr.c | 29 +-- net/xfrm/xfrm_user.c | 6 +- security/capability.c | 3 +- security/security.c | 6 +- security/selinux/hooks.c | 13 +- security/selinux/include/security.h | 2 +- security/selinux/include/xfrm.h | 3 +- security/selinux/selinuxfs.c | 28 ++- security/selinux/ss/services.c | 6 +- security/selinux/xfrm.c | 14 +- sound/core/compress_offload.c | 2 +- sound/pci/oxygen/xonar_dg.c | 30 ++- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/fsl_esai.c | 2 + sound/soc/fsl/fsl_utils.c | 27 +++ sound/soc/fsl/fsl_utils.h | 4 +- sound/soc/fsl/imx-ssi.c | 2 + sound/soc/generic/simple-card.c | 220 +++++++++++++-------- sound/soc/sh/rcar/src.c | 18 +- sound/soc/soc-core.c | 17 +- sound/soc/tegra/tegra20_ac97.c | 15 +- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra_wm9712.c | 17 +- tools/perf/bench/numa.c | 1 + tools/perf/builtin-bench.c | 2 +- 125 files changed, 1152 insertions(+), 840 deletions(-)

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[alsa-devel] [PATCH 1/2] snd-ice1712: add suspend support for ICE1712 chip
by Ondrej Zary 31 Mar '14

31 Mar '14

Add suspend/resume support for ICE1712 chip. Card-specific subdrivers need to enable it and provide callbacks that suspend/resume the codecs. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ice1712/ice1712.c | 85 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 77 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 28ec872..dbb0f96 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2428,6 +2428,13 @@ static int snd_ice1712_chip_init(struct snd_ice1712 *ice) snd_ice1712_write(ice, ICE1712_IREG_CONSUMER_POWERDOWN, 0); } snd_ice1712_set_pro_rate(ice, 48000, 1); + /* unmask used interrupts */ + outb(((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) == 0 ? + ICE1712_IRQ_MPU2 : 0) | + ((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97) ? + ICE1712_IRQ_PBKDS | ICE1712_IRQ_CONCAP | ICE1712_IRQ_CONPBK : 0), + ICEREG(ice, IRQMASK)); + outb(0x00, ICEMT(ice, IRQ)); return 0; } @@ -2589,6 +2596,7 @@ static int snd_ice1712_create(struct snd_card *card, ice->pci = pci; ice->irq = -1; pci_set_master(pci); + /* disable legacy emulation */ pci_write_config_word(ice->pci, 0x40, 0x807f); pci_write_config_word(ice->pci, 0x42, 0x0006); snd_ice1712_proc_init(ice); @@ -2625,14 +2633,6 @@ static int snd_ice1712_create(struct snd_card *card, return -EIO; } - /* unmask used interrupts */ - outb(((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) == 0 ? - ICE1712_IRQ_MPU2 : 0) | - ((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97) ? - ICE1712_IRQ_PBKDS | ICE1712_IRQ_CONCAP | ICE1712_IRQ_CONPBK : 0), - ICEREG(ice, IRQMASK)); - outb(0x00, ICEMT(ice, IRQ)); - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); if (err < 0) { snd_ice1712_free(ice); @@ -2809,11 +2809,80 @@ static void snd_ice1712_remove(struct pci_dev *pci) snd_card_free(card); } +#ifdef CONFIG_PM_SLEEP +static int snd_ice1712_suspend(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(ice->pcm); + snd_pcm_suspend_all(ice->pcm_pro); + snd_pcm_suspend_all(ice->pcm_ds); + snd_ac97_suspend(ice->ac97); + + if (ice->pm_suspend) + ice->pm_suspend(ice); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, PCI_D3hot); + return 0; +} + +static int snd_ice1712_resume(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + pci_set_master(pci); + + if (snd_ice1712_chip_init(ice) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + if (ice->pm_resume) + ice->pm_resume(ice); + + if (ice->ac97) + snd_ac97_resume(ice->ac97); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static SIMPLE_DEV_PM_OPS(snd_ice1712_pm, snd_ice1712_suspend, snd_ice1712_resume); +#define SND_VT1712_PM_OPS &snd_ice1712_pm +#else +#define SND_VT1712_PM_OPS NULL +#endif /* CONFIG_PM_SLEEP */ + static struct pci_driver ice1712_driver = { .name = KBUILD_MODNAME, .id_table = snd_ice1712_ids, .probe = snd_ice1712_probe, .remove = snd_ice1712_remove, + .driver = { + .pm = SND_VT1712_PM_OPS, + }, }; module_pci_driver(ice1712_driver); -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: rsnd: fix src clock prepare/unprepare
by Ben Dooks 31 Mar '14

31 Mar '14

As with the previous commit, before a clock can be used it must be prepared for use. Change from clk_enable() and clk_disable() to the versions of the calls which also prepare and un-prepare the clocks. Will fix warnings from the clock code when this is used. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/scu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 9153a11..3b1d1d2 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -284,7 +284,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } - clk_enable(scu->clk); + clk_prepare_enable(scu->clk); /* it use DMA transter */ @@ -317,7 +317,7 @@ static int rsnd_scu_stop(struct rsnd_mod *mod, rsnd_scu_transfer_stop(priv, mod, rdai, io); - clk_disable(scu->clk); + clk_disable_unprepare(scu->clk); return 0; } -- 1.9.0

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Re: [alsa-devel] is 440 Hz the true harmonic wave
by Joël Krähemann 31 Mar '14

31 Mar '14

On Mon, 2014-03-31 at 10:27 +0200, Felix Homann wrote: > 2014-03-31 10:20 GMT+02:00 Joël Krähemann <joel(a)weedlight.ch>: > > > I believe that 440 Hz is an approximation rather than true > harmonic on > it's pressure. Or am I wrong? Sine is harmonic in default arc. > > > > What are you talking about??? About scale. Am I wrong when I believe 440 Hz has physical property to be harmonic?

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[alsa-devel] [PATCH] ALSA: hda - Enable beep for ASUS 1015E
by W. Trevor King 31 Mar '14

31 Mar '14

The `lspci -nnvv` output contains (wrapped for line length): 00:1b.0 Audio device [0403]: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller [8086:1e20] (rev 04) Subsystem: ASUSTeK Computer Inc. Device [1043:115d] Signed-off-by: W. Trevor King <wking(a)tremily.us> --- This patch is against git://git.alsa-project.org/alsa-kernel.git master (currently e38d6d3, ALSA: hda - Add Dell headset detection quirk for three laptop models, 2013-12-13), but I imagine it will apply cleanly to any recent branch. I'm happy to fill in any other information folks want, in case there's uncertainty about whether it's just a one-off quirk for my hardware instance, or if it's a general quirk for all 1043:115d devices. Cheers, Trevor sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3578f11..7cfe209 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1001,6 +1001,7 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x115d, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), -- 1.9.1.353.gc66d89d

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[alsa-devel] [patch] ALSA: asihpi: fix some indenting in snd_card_asihpi_pcm_new()
by Dan Carpenter 31 Mar '14

31 Mar '14

This used to be a part of a condition until f3d145aac913 ('ALSA: asihpi: MMAP for non-busmaster cards') but now it's not and we can remove an indent level. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e4e42f2..901c949 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1253,11 +1253,12 @@ static int snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, int device) num_outstreams, num_instreams, &pcm); if (err < 0) return err; + /* pointer to ops struct is stored, dont change ops afterwards! */ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_card_asihpi_playback_mmap_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_asihpi_capture_mmap_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_card_asihpi_playback_mmap_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_card_asihpi_capture_mmap_ops); pcm->private_data = asihpi; pcm->info_flags = 0;

2 1

[alsa-devel] is 440 Hz the true harmonic wave
by Joël Krähemann 31 Mar '14

31 Mar '14

Is 440 Hz the true harmonic wave? Is there a constant for it in glibc, like M_PI?

2 2

[alsa-devel] RT priority 99 on soundcard
by Joël Krähemann 31 Mar '14

31 Mar '14

Is running on top of soundcard with realtime priority 99 problematic? /* and now async */ { static const struct timespec req = { 0, (250000000 * (1 / 45)), }; if(!AGS_IS_DEVOUT_THREAD(thread)){ nanosleep(&req, NULL); } }

2 1

[alsa-devel] [PATCH 1/6] ASoC: rt5640: Remove the pre-allocated size of reg_default
by Oder Chiou 31 Mar '14

31 Mar '14

In order to prevent the redundant memory usage, the pre-allocated size of reg_default should be remove. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index cf041e3..9427511 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -59,7 +59,7 @@ static struct reg_default init_list[] = { }; #define RT5640_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5640_reg[RT5640_VENDOR_ID2 + 1] = { +static const struct reg_default rt5640_reg[] = { { 0x00, 0x000e }, { 0x01, 0xc8c8 }, { 0x02, 0xc8c8 }, -- 1.8.1.1.439.g50a6b54

2 12

[alsa-devel] [PATCH] ASoC: rsnd: use fast_io for bus
by Ben Dooks 29 Mar '14

29 Mar '14

The rcar sound driver uses regmap to access registers in various parts of the block and uses regmap to manage mappings. The regmap is created without fast_io set, which means it locks with a mutex rather than a lighter-spinlock. The use of the mutex lock causes issues when the IRQ handler is entered as the code needs to read/write register values and thus with lock debugging enabled the system outputs a number of warnings such as: BUG: sleeping function called from invalid context at rnel/locking/mutex.c:616 The rcar registers are all connected via APB bus and thus not that slow to access. The fix is to set the fast_io in the code so that regmap creates the regmap structures using the faster spinlock functions. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/gen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 9094970..6468962 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -66,6 +66,7 @@ static int rsnd_regmap_read32(void *context, } static struct regmap_bus rsnd_regmap_bus = { + .fast_io = true, .write = rsnd_regmap_write32, .read = rsnd_regmap_read32, .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, -- 1.9.0

2 1

[alsa-devel] Input/output error Alsa api
by Thiago Carísio 29 Mar '14

29 Mar '14

Hi! I am Thiago and i'm starting to use the alsa api. I'm testing one function to make the sound capture and i got this input/output error. I don't get it why this is happening...so, please...some kind soul can help me? Thanks

1 0

[alsa-devel] Advanced Gtk+ Sequencer - Beta Release
by Joël Krähemann 29 Mar '14

29 Mar '14

Hi, I just released the very first beta, 3 hours ago. There are 2 weeks towards final release 0.4.0 and any suggestions are welcome. https://sourceforge.net/projects/ags/

1 0

Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 28 Mar '14

28 Mar '14

> > On 03/26/2014 11:41 PM, Songhee Baek wrote: > > >> On 03/26/2014 01:02 AM, Arun Shamanna Lakshmi wrote: > > >> > > >> The way you describe this it seems to me that a value array for > > >> this kind of mux would look like. > > >> > > >> 0x00000000, 0x00000000, 0x00000001 > > >> 0x00000000, 0x00000000, 0x00000002 > > >> 0x00000000, 0x00000000, 0x00000003 > > >> 0x00000000, 0x00000000, 0x00000004 > > >> 0x00000000, 0x00000000, 0x00000008 > > >> ... > > >> > > >> That seems to be extremely tedious. If the MUX uses a one hot > > >> encoding how about storing the index of the bit in the values array > > >> and use (1 << value) when writing the value to the register? > > > > > > If we store the index of the bit, the value will be duplicated for > > > each > > registers inputs since register has 0 to 31bits to shift, then we need > > to decode the index to interpret value for which registers to set. If > > we need to interpret the decoded value of index, it is better to have > > custom put/get function in our driver, isn't it? > > > > > > > I'm not sure I understand. If you use (val / 32) to pick the register > > and (val % > > 32) to pick the bit in the register this should work just fine. Maybe > > I'm missing something. Do you have a real world code example of of the > > this type of enum is used? > > > > I can use val/32 and val%32 for this multi register mux. With this logic, put function would be easy however get function becomes tedious due to looping on each bit per register for 3 of them in our case. Rather, it would be easy to identify a unique MUX_OFFSET to distinguish between the 3 registers as shown in the code below. #define MULTI_MUX_INPUT_OFFSET(n) (5 * n) int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, reg_idx; if (e->reg[0] != SND_SOC_NOPM) { for (reg_idx = 0; reg_idx < e->num_regs; reg_idx) { reg_val = snd_soc_read(codec, e->reg[reg_idx]); val = (reg_val >> e->shift_l) & e->mask[reg_idx]; if (val) val += MULTI_MUX_INPUT_OFFSET(reg_idx); } } else { reg_val = dapm_kcontrol_get_value(kcontrol); val = (reg_val >> e->shift_l) & e->mask[0]; } ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } return 0; } int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int change, i, value, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; value = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; value |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = value - MULTI_MUX_INPUT_OFFSET(i); /* checking reg_val is power of 2 : one-hot code */ /* if reg_val after subtract MULTI_MUX_INPUT_OFFSET is not power of 2, reg[i] should be zero */ if (reg_val & (reg_val - 1)) { /* clear the current input register */ snd_soc_write(codec, e->reg[i], 0); } else { /* reg_val is power of 2, store updated info */ value = reg_val; mask = e->mask[i]; update_idx = i; } } update_reg: mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg[update_idx] != SND_SOC_NOPM) change = snd_soc_test_bits(codec, e->reg[update_idx], mask, value); else change = dapm_kcontrol_set_value(kcontrol, value); if (change) { if (e->reg[update_idx] != SND_SOC_NOPM) { update.kcontrol = kcontrol; update.reg = e->reg[update_idx]; update.mask= mask; update.val = value; card->update = &update; } ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); return change; } > > >>> - int reg; > > >>> + int reg[SOC_ENUM_MAX_REGS]; > > >>> unsigned char shift_l; > > >>> unsigned char shift_r; > > >>> unsigned int items; > > >>> - unsigned int mask; > > >>> + unsigned int mask[SOC_ENUM_MAX_REGS]; > > >> > > >> If you make mask and reg pointers instead of arrays this should be > > >> much more flexible and not be limited to 3 registers. We will make reg* and mask* instead of arrays and since we use the same structure, the plan is to share the get and put function code. Thanks. Songhee.

1 0

[alsa-devel] [PATCH] ALSA: hda - verify pin:cvt connection on preparing a stream for Intel HDMI codec
by mengdong.lin＠intel.com 28 Mar '14

28 Mar '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The pin:converter connection may get reset after S3 on Intel Broadwell HDMI codec. And this can cause multiple pins share a same convertor and mute control will affect each other. We observed a resumed audio playback become silent after S3. This patch verifies pin:cvt connection on preparing a stream, to assure the pin selects the right convetor and an assigned convertor is not shared by other unused pins. Apply this fix-up on Haswell, Broadwell and Baytrail. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3ab7063..6000ce9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -68,6 +68,7 @@ struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; + int mux_idx; hda_nid_t cvt_nid; struct hda_codec *codec; @@ -1342,6 +1343,8 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_idx == spec->num_cvts) return -ENODEV; + per_pin->mux_idx = mux_idx; + if (cvt_id) *cvt_id = cvt_idx; if (mux_id) @@ -1350,6 +1353,22 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } +/* Assure the pin select the right convetor */ +static void intel_verify_pin_cvt_connect(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + hda_nid_t pin_nid = per_pin->pin_nid; + int mux_idx, curr; + + mux_idx = per_pin->mux_idx; + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_idx) + snd_hda_codec_write_cache(codec, pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); +} + /* Intel HDMI workaround to fix audio routing issue: * For some Intel display codecs, pins share the same connection list. * So a conveter can be selected by multiple pins and playback on any of these @@ -1751,6 +1770,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; + if (is_haswell_plus(codec) || is_valleyview(codec)) { + /* the pin:cvt connection may get reset after S3 */ + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); + } + non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); per_pin->channels = substream->runtime->channels; -- 1.8.1.2

4 10

Re: [alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by Raymond Yau 28 Mar '14

28 Mar '14

> > I've previously posted under ALSA's user mailing list so to avoid the > clutter, the details about what's happening is here: > http://article.gmane.org/gmane.linux.alsa.user/38258 > http://article.gmane.org/gmane.linux.alsa.user/38260 > > There was no debug info under /proc/asound/card0/ so nothing to give from > there unless you guys need something specific. > > Output of alsa-debug.sh: > http://www.alsa-project.org/db/?f=26187faadd47f3626029ba667cb004e74695f2b2 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem cannot read valid value from register within three trials state.CMI8738 { control.1 { iface PCM device 2 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 comment { access read type INTEGER count 6 range '0 - 36' } } Seem bug in adding channel map to iec958 instead of analog https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… pcm.!default { type plug slave.pcm "dmixer" } You need asym plugin for playback and capture when overwrite pcm.default

3 5

[alsa-devel] tlv320aic32x4 Codec ADC and DAC must shutdown to alter clocks
by Mike Looijmans 28 Mar '14

28 Mar '14

My general question: If "my" codec needs to alter the state of the DAPM tree, how can the codec ask or inform the DAPM components of this? This problem occurs on tlv320aic32x4 codecs, but probably applies to many others as well. Using the tlv320aic32x4 a simple loop like this may "crash" the codec: while true do speaker-test -c 2 -t sine -l 1 -r 48000 > /dev/null speaker-test -c 2 -t sine -l 1 -r 44100 > /dev/null done A "crash" means, in this respect that the chip loses some of its settings, malfunctions and often it results in NACKs on I2C commands. I've been trying to properly fix this, but it just won't fit in the framework. The cause is that the DAPM starts the DAC and/or ADC cores while the clocks aren't running, and stops the clocks before shutting down the cores. There is the overall bug of the driver that starts and stops all the clocks in the aic32x4_set_bias_level in the wrong order - it should start them from source to sink, and stop them in the opposite direction, so the PLL must be the first to start and last to stop. Apparently this isn't really critical as I've never observed failure as a result of this. What I cannot fix is that changing the parameters (like clock frequency) must stop the DAC core. The DAPM framework just sets the BIAS to "ON" only once, and does not shut down until after a time out. Which is good. But if the next stream to play needs a different clock setting, the DAC must be stopped first, the code must wait until the DAC has powered down, and then the clocks can be stopped and altered. When all settings are done, the DAC can be restarted. Starting the DAC is the DAPM's task, it's this line that does the job: SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), I can detect the change in the hw_params, and on the first stream to play, this is called while still in STANDBY or lower levels. However, when the codec is already in ON bias level, how do I tell DAPM that I had to shut down the DAC in order to change the parameters? Just toggling the related bits will keep the DAC powered down. A routine to store and restore the power state is possible, but very ineffective. Especially since the "I must shutdown" conclusion can also be in the clock setting routine. That would result in multiple down-up cycles. -- Mike Looijmans

2 5

[alsa-devel] [PATCH v6 0/2] ASoC: Add SiRF I2S controller and hdmi card drivers
by RongJun Ying 28 Mar '14

28 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver Rongjun Ying (2): ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi support .../devicetree/bindings/sound/sirf-i2s.txt | 27 ++ sound/soc/sirf/Kconfig | 11 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-i2s.c | 445 ++++++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 93 ++++ 5 files changed, 578 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h -- 1.7.5.4

2 4

[alsa-devel] [PATCH] ASoC: fsl_sai: Improve fsl_sai_isr()
by Nicolin Chen 28 Mar '14

28 Mar '14

This patch improves fsl_sai_isr() in these ways: 1, Add comment for mask fetching code. 2, Return IRQ_NONE if the IRQ is not for the device. 3, Use regmap_write() instead of regmap_update_bits(). Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 64 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 45 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0bc98bb..f088545 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -30,62 +30,88 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) { struct fsl_sai *sai = (struct fsl_sai *)devid; struct device *dev = &sai->pdev->dev; - u32 xcsr, mask; + u32 flags, xcsr, mask; + bool irq_none = true; - /* Only handle those what we enabled */ + /* + * Both IRQ status bits and IRQ mask bits are in the xCSR but + * different shifts. And we here create a mask only for those + * IRQs that we activated. + */ mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT; /* Tx IRQ */ regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); - xcsr &= mask; + flags = xcsr & mask; - if (xcsr & FSL_SAI_CSR_WSF) + if (flags) + irq_none = false; + else + goto irq_rx; + + if (flags & FSL_SAI_CSR_WSF) dev_dbg(dev, "isr: Start of Tx word detected\n"); - if (xcsr & FSL_SAI_CSR_SEF) + if (flags & FSL_SAI_CSR_SEF) dev_warn(dev, "isr: Tx Frame sync error detected\n"); - if (xcsr & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) { dev_warn(dev, "isr: Transmit underrun detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } - if (xcsr & FSL_SAI_CSR_FWF) + if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); - if (xcsr & FSL_SAI_CSR_FRF) + if (flags & FSL_SAI_CSR_FRF) dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + flags &= FSL_SAI_CSR_xF_W_MASK; + xcsr &= ~FSL_SAI_CSR_xF_MASK; + + if (flags) + regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); +irq_rx: /* Rx IRQ */ regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); - xcsr &= mask; + flags = xcsr & mask; - if (xcsr & FSL_SAI_CSR_WSF) + if (flags) + irq_none = false; + else + goto out; + + if (flags & FSL_SAI_CSR_WSF) dev_dbg(dev, "isr: Start of Rx word detected\n"); - if (xcsr & FSL_SAI_CSR_SEF) + if (flags & FSL_SAI_CSR_SEF) dev_warn(dev, "isr: Rx Frame sync error detected\n"); - if (xcsr & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) { dev_warn(dev, "isr: Receive overflow detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } - if (xcsr & FSL_SAI_CSR_FWF) + if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); - if (xcsr & FSL_SAI_CSR_FRF) + if (flags & FSL_SAI_CSR_FRF) dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + flags &= FSL_SAI_CSR_xF_W_MASK; + xcsr &= ~FSL_SAI_CSR_xF_MASK; - return IRQ_HANDLED; + if (flags) + regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); + +out: + if (irq_none) + return IRQ_NONE; + else + return IRQ_HANDLED; } static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, -- 1.8.4

2 1

[alsa-devel] [PATCH 1/3 v2] ASoC: ak5386: add regulator consumer support
by Daniel Mack 28 Mar '14

28 Mar '14

The chip has two power supplies, VA and VDD. Enable them both as long as the codec is in use. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak5386.c | 50 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b..a30be5c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char const *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { { "Capture", NULL, "AINR" }, }; +static int ak5386_soc_probe(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int ak5386_soc_remove(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +#ifdef CONFIG_PM +static int ak5386_soc_suspend(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int ak5386_soc_resume(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define ak5386_soc_suspend NULL +#define ak5386_soc_resume NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver soc_codec_ak5386 = { + .probe = ak5386_soc_probe, + .remove = ak5386_soc_remove, + .suspend = ak5386_soc_suspend, + .resume = ak5386_soc_resume, .dapm_widgets = ak5386_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), .dapm_routes = ak5386_dapm_routes, @@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); -- 1.8.5.3

2 9

[alsa-devel] [PATCH v2 0/4] ASoC: davinci-mcasp: Correct start/stop sequences
by Peter Ujfalusi 28 Mar '14

28 Mar '14

Hi, Changes since v1: - Correction for the stop sequence is added - start code cleanup patch added The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. When stopping the stream the AFIFO should be stopped as the last step to avoid unpredictable behavior. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions sound/soc/davinci/davinci-mcasp.c | 95 ++++++++++++++++++--------------------- sound/soc/davinci/davinci-mcasp.h | 6 +++ 2 files changed, 49 insertions(+), 52 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: sta350: Add codec driver
by Daniel Mack 28 Mar '14

28 Mar '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: * Added my S-o-b * Made the Kconfig symbol visible * Documented regulator notes in DT bindings doc * Fixed spelling and other style nits * Converted to SOC_ENUM_SINGLE_DECL * Added a lock around the logic in sta350_coefficient_get() * Added a DAPM route from "DAC" to "Playback" * Use dev_dbg() rather than pr_debug() * Changed the code to make use of regmap_update_bits rather than open-coding the same functionality * Cleaned up the reset sequence logic * If a gpio was requested, the claiming has to succeed (handles EPROBE_DEFER cases) * Kick codec->control_data initialization * Skip initialization of default values in regcache * Apply power before running init sequences * Parse the DT without consulting of_match_device() .../devicetree/bindings/sound/st,sta350.txt | 107 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1266 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1660 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..07f9c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,107 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpio: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpio: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpio = <&gpio1 19 0>; + power-down-gpio = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * 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. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..c7b853f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,10 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate "STA350 speaker amplifier" + depends on I2C + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..915b8769 --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1266 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.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. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.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 <sound/tlv.h> + +#include <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char const *sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + int gpio_nreset; + int gpio_power_down; + + struct mutex coeff_lock; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char const *sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char const *sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char const *sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char const *sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char const *sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char const *sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char const *sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char const *sta350_noise_shaper_type[] = { + "Third order", "Fourth order" +}; + +static const unsigned int sta350_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta350_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta350_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta350_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static SOC_ENUM_SINGLE_DECL(sta350_drc_ac_enum, + STA350_CONFD, STA350_CONFD_DRC_SHIFT, + sta350_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta350_noise_shaper_enum, + STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + sta350_noise_shaper_type); +static SOC_ENUM_SINGLE_DECL(sta350_auto_gc_enum, + STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + sta350_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta350_auto_xo_enum, + STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + sta350_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch1_enum, + STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch2_enum, + STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch3_enum, + STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch1_enum, + STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch2_enum, + STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch3_enum, + STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_attack_rate_enum, + STA350_L1AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_attack_rate_enum, + STA350_L2AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_release_rate_enum, + STA350_L1AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_release_rate_enum, + STA350_L2AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i, ret = 0; + + mutex_lock(&sta350->coeff_lock); + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + } else if (numcoef == 5) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + } else { + ret = -EINVAL; + goto exit_unlock; + } + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + +exit_unlock: + mutex_unlock(&sta350->coeff_lock); + + return ret; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, + { "DAC", NULL, "Playback" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_C1IM | STA350_CONFB_C2IM, confb); +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + int ret; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + dev_dbg(codec->dev, "rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + } + + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) { + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + } + + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + confa = (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + confb = 0; + + switch (params_width(params)) { + case 24: + dev_dbg(codec->dev, "24bit\n"); + /* fall through */ + case 32: + dev_dbg(codec->dev, "24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case 20: + dev_dbg(codec->dev, "20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case 18: + dev_dbg(codec->dev, "18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case 16: + dev_dbg(codec->dev, "16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK, + confa); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB, + confb); + if (ret < 0) + return ret; + + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 1); + + if (gpio_is_valid(sta350->gpio_nreset)) { + gpio_set_value(sta350->gpio_nreset, 0); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + + /* power down: low */ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + if (gpio_is_valid(sta350->gpio_nreset)) + gpio_set_value(sta350->gpio_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char const *sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + sta350->gpio_nreset = of_get_named_gpio(np, "reset-gpio", 0); + sta350->gpio_power_down = of_get_named_gpio(np, "power-down-gpio", 0); + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(&i2c->dev, sizeof(struct sta350_priv), + GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + sta350->pdata = dev_get_platdata(&i2c->dev); + sta350->gpio_nreset = -EINVAL; + sta350->gpio_power_down = -EINVAL; + + mutex_init(&sta350->coeff_lock); + +#ifdef CONFIG_OF + ret = sta350_probe_dt(&i2c->dev, sta350); + if (ret < 0) + return ret; +#endif + + /* GPIOs */ + if (gpio_is_valid(sta350->gpio_nreset)) { + ret = devm_gpio_request_one(&i2c->dev, sta350->gpio_nreset, + GPIOF_OUT_INIT_HIGH, + "ST350 Reset"); + if (ret < 0) + return ret; + } + + if (gpio_is_valid(sta350->gpio_power_down)) { + ret = devm_gpio_request_one(&i2c->dev, sta350->gpio_power_down, + GPIOF_OUT_INIT_HIGH, + "ST350 Power-Down"); + if (ret < 0) + return ret; + } + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(&i2c->dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.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. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

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[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Correct start sequences for TX/RX
by Peter Ujfalusi 28 Mar '14

28 Mar '14

Hi, The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++------------------------ sound/soc/davinci/davinci-mcasp.h | 6 ++++++ 2 files changed, 19 insertions(+), 24 deletions(-) -- 1.9.1

1 3

[alsa-devel] [PATCH v2] ASoC: fsl_sai: Add isr to deal with error flag
by Nicolin Chen 28 Mar '14

28 Mar '14

It's quite cricial to clear error flags because SAI might hang if getting FIFO underrun during playback (I haven't confirmed the same issue on Rx overflow though). So this patch enables those irq and adds isr() to clear the flags so as to keep playback entirely safe. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Mask the active flags only for the following handler. * Reset FIFO for FIFO underrun/overflow cases. * Enable two error flags only as default. * Use dev_warn for two error flags. * Only clear those W1C bits. sound/soc/fsl/fsl_sai.c | 85 +++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 15 +++++++++ 2 files changed, 97 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4a4231..0bc98bb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,71 @@ #include "fsl_sai.h" +#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ + FSL_SAI_CSR_FEIE) + +static irqreturn_t fsl_sai_isr(int irq, void *devid) +{ + struct fsl_sai *sai = (struct fsl_sai *)devid; + struct device *dev = &sai->pdev->dev; + u32 xcsr, mask; + + /* Only handle those what we enabled */ + mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT; + + /* Tx IRQ */ + regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Tx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Tx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Transmit underrun detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + /* Rx IRQ */ + regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); + xcsr &= mask; + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Rx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_warn(dev, "isr: Rx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) { + dev_warn(dev, "isr: Receive overflow detected\n"); + /* FIFO reset for safety */ + xcsr |= FSL_SAI_CSR_FR; + } + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); + + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_xF_W_MASK | FSL_SAI_CSR_FR, xcsr); + + return IRQ_HANDLED; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -373,8 +438,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -490,12 +555,14 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int ret; + int irq, ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->pdev = pdev; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -514,6 +581,18 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(sai->regmap); } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e432260..a264185 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -37,7 +37,21 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_xF_SHIFT 16 +#define FSL_SAI_CSR_xF_W_SHIFT 18 +#define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) +#define FSL_SAI_CSR_xF_W_MASK (0x7 << FSL_SAI_CSR_xF_W_SHIFT) +#define FSL_SAI_CSR_WSF BIT(20) +#define FSL_SAI_CSR_SEF BIT(19) +#define FSL_SAI_CSR_FEF BIT(18) #define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRF BIT(16) +#define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_WSIE BIT(12) +#define FSL_SAI_CSR_SEIE BIT(11) +#define FSL_SAI_CSR_FEIE BIT(10) +#define FSL_SAI_CSR_FWIE BIT(9) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) @@ -99,6 +113,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { + struct platform_device *pdev; struct regmap *regmap; bool big_endian_regs; -- 1.8.4

2 2

[alsa-devel] [PATCH] sound: rcar: use fast_io for bus
by Ben Dooks 27 Mar '14

27 Mar '14

The rcar sound driver uses regmap to access registers in various parts of the block and uses regmap to manage mappings. The regmap is created without fast_io set, which means it locks with a mutex rather than a lighter-spinlock. The use of the mutex lock causes issues when the IRQ handler is entered as the code needs to read/write register values and thus with lock debugging enabled the system outputs a number of warnings such as: BUG: sleeping function called from invalid context at rnel/locking/mutex.c:616 The rcar registers are all connected via APB bus and thus not that slow to access. The fix is to set the fast_io in the code so that regmap creates the regmap structures using the faster spinlock functions. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/gen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 9094970..6468962 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -66,6 +66,7 @@ static int rsnd_regmap_read32(void *context, } static struct regmap_bus rsnd_regmap_bus = { + .fast_io = true, .write = rsnd_regmap_write32, .read = rsnd_regmap_read32, .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: sta350: Add codec driver
by Daniel Mack 27 Mar '14

27 Mar '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> --- I'm sending this patch on behalf of Sven Brandau, who wrote the driver, based on earlier bits for the STA320. In case of any pending issue, I will amend and resubmit. Thanks, Daniel .../devicetree/bindings/sound/st,sta350.txt | 103 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1277 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1666 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..ae0630b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,103 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpio: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpio: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpio = <&gpio1 19 0>; + power-down-gpio = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * 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. + */ +#ifndef __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..6211889 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,9 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..c78d1c5 --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1277 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.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. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.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 <sound/tlv.h> + +#include <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char const *sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + int gpio_nreset; + int gpio_power_down; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char const *sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char const *sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char const *sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char const *sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char const *sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char const *sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char const *sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char const *sta350_noise_shaper_type[] = { + "Third order", "Forth order" +}; + +static const unsigned int sta350_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta350_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta350_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta350_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta350_drc_ac_enum = + SOC_ENUM_SINGLE(STA350_CONFD, STA350_CONFD_DRC_SHIFT, + 2, sta350_drc_ac); +static const struct soc_enum sta350_noise_shaper_enum = + SOC_ENUM_SINGLE(STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + 2, sta350_noise_shaper_type); +static const struct soc_enum sta350_auto_gc_enum = + SOC_ENUM_SINGLE(STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + 4, sta350_auto_gc_mode); +static const struct soc_enum sta350_auto_xo_enum = + SOC_ENUM_SINGLE(STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + 16, sta350_auto_xo_mode); +static const struct soc_enum sta350_binary_output_ch1_enum = + SOC_ENUM_SINGLE(STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_binary_output_ch2_enum = + SOC_ENUM_SINGLE(STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_binary_output_ch3_enum = + SOC_ENUM_SINGLE(STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + 2, sta350_binary_output); +static const struct soc_enum sta350_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + 3, sta350_limiter_select); +static const struct soc_enum sta350_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA350_L1AR, STA350_LxA_SHIFT, + 16, sta350_limiter_attack_rate); +static const struct soc_enum sta350_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA350_L2AR, STA350_LxA_SHIFT, + 16, sta350_limiter_attack_rate); +static const struct soc_enum sta350_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA350_L1AR, STA350_LxR_SHIFT, + 16, sta350_limiter_release_rate); +static const struct soc_enum sta350_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA350_L2AR, STA350_LxR_SHIFT, + 16, sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + else + return -EINVAL; + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + + return 0; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + pr_debug("mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb; + + regmap_read(sta350->regmap, STA350_CONFB, &confb); + + pr_debug("\n"); + confb &= ~(STA350_CONFB_C1IM | STA350_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + regmap_write(sta350->regmap, STA350_CONFB, confb); + return 0; +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + pr_debug("rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + regmap_read(sta350->regmap, STA350_CONFA, &confa); + confa &= ~(STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK); + confa |= (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + + regmap_read(sta350->regmap, STA350_CONFB, &confb); + confb &= ~(STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + regmap_write(sta350->regmap, STA350_CONFA, confa); + regmap_write(sta350->regmap, STA350_CONFB, confb); + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 1); + + if (gpio_is_valid(sta350->gpio_nreset)) { + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 0); + mdelay(1); + gpio_set_value(sta350->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + /* power down: low */ + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + if (gpio_is_valid(sta350->gpio_nreset)) + if (devm_gpio_request_one(codec->dev, sta350->gpio_nreset, + GPIOF_OUT_INIT_HIGH, + "ST350 Reset")) + sta350->gpio_nreset = -EINVAL; + + if (gpio_is_valid(sta350->gpio_power_down)) + if (devm_gpio_request_one(codec->dev, sta350->gpio_power_down, + GPIOF_OUT_INIT_HIGH, + "ST350 Power-Down")) + sta350->gpio_power_down = -EINVAL; + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* + * Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + codec->control_data = sta350->regmap; + + /* + * Chip documentation explicitly requires that the reset values + * of reserved register bits are left untouched. + * Write the register default value to cache for reserved registers, + * so the write to the these registers are suppressed by the cache + * restore code when it skips writes of default registers. + */ + regcache_cache_only(sta350->regmap, true); + regmap_write(sta350->regmap, STA350_CONFC, 0xc2); + regmap_write(sta350->regmap, STA350_CONFE, 0xc2); + regmap_write(sta350->regmap, STA350_CONFF, 0x5c); + regmap_write(sta350->regmap, STA350_MMUTE, 0x10); + regmap_write(sta350->regmap, STA350_AUTO1, 0x60); + regmap_write(sta350->regmap, STA350_AUTO3, 0x00); + regmap_write(sta350->regmap, STA350_C3CFG, 0x40); + regcache_cache_only(sta350->regmap, false); + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(sta350->gpio_nreset)) + gpio_set_value(sta350->gpio_nreset, 0); + + if (gpio_is_valid(sta350->gpio_power_down)) + gpio_set_value(sta350->gpio_power_down, 0); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char const *sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + sta350->gpio_nreset = of_get_named_gpio(np, "reset-gpio", 0); + sta350->gpio_power_down = of_get_named_gpio(np, "power-down-gpio", 0); + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(&i2c->dev, sizeof(struct sta350_priv), + GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + sta350->pdata = dev_get_platdata(&i2c->dev); + sta350->gpio_nreset = -EINVAL; + +#ifdef CONFIG_OF + if (of_match_device(st350_dt_ids, &i2c->dev)) { + ret = sta350_probe_dt(&i2c->dev, sta350); + if (ret < 0) + return ret; + } +#endif + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(&i2c->dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.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. + */ +#ifndef _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Arun Shamanna Lakshmi 27 Mar '14

27 Mar '14

If the mux uses 1 bit position per input, and requires to set one single bit at a time, then an N bit register can support up to N inputs. In more recent Tegra chips, we have at least greater than 64 inputs which requires at least 2 .reg fields in struct soc_enum. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 20 +++++- include/sound/soc.h | 44 ++++++++++--- sound/soc/soc-core.c | 118 +++++++++++++++++++++++++++++++++-- sound/soc/soc-dapm.c | 155 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 300 insertions(+), 37 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..324de75 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -315,6 +315,12 @@ struct device; .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) +#define SOC_DAPM_VALUE_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_wide, \ + .get = snd_soc_dapm_get_value_enum_wide, \ + .put = snd_soc_dapm_put_value_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_PIN_SWITCH(xname) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \ .info = snd_soc_dapm_info_pin_switch, \ @@ -352,6 +358,9 @@ struct device; /* regulator widget flags */ #define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* maximum number of registers to update */ +#define SOC_DAPM_UPDATE_MAX_REGS 3 + struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; @@ -378,6 +387,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +603,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[SOC_DAPM_UPDATE_MAX_REGS]; + int mask[SOC_DAPM_UPDATE_MAX_REGS]; + int val[SOC_DAPM_UPDATE_MAX_REGS]; + int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..67097c6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,16 +177,22 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg[0] = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask[0] = xitems ? roundup_pow_of_two(xitems) - 1 : 0, .num_regs = 1 } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } +#define SOC_VALUE_ENUM_TRIPLE(reg1, reg2, reg3, mask1, mask2, mask3, \ + nreg, nmax, xtexts, xvalues) \ +{ .reg[0] = reg1, .reg[1] = reg2, .reg[2] = reg3, \ + .mask[0] = mask1, .mask[1] = mask2, .mask[2] = mask3, \ + .num_regs = nreg, .items = nmax, .texts = xtexts, .values = xvalues } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg[0] = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask[0] = xmask, .items = xitems, .texts = xtexts, .values = xvalues,\ + .num_regs = 1 } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ @@ -198,6 +204,12 @@ .private_value = (unsigned long)&xenum } #define SOC_VALUE_ENUM(xname, xenum) \ SOC_ENUM(xname, xenum) +#define SOC_VALUE_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ + .info = snd_soc_info_enum_wide, \ + .get = snd_soc_get_value_enum_wide, \ + .put = snd_soc_put_value_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -297,7 +309,13 @@ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts) - +#define SOC_VALUE_ENUM_TRIPLE_DECL(name, reg1, reg2, reg3, \ + mask1, mask2, mask3, \ + xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_TRIPLE(reg1, reg2, reg3, \ + mask1, mask2, mask3, 3, \ + ARRAY_SIZE(xtexts), \ + xtexts, xvalues) /* * Component probe and remove ordering levels for components with runtime * dependencies. @@ -309,6 +327,11 @@ #define SND_SOC_COMP_ORDER_LAST 2 /* + * The maximum number of registers in soc_enum + */ +#define SOC_ENUM_MAX_REGS 3 + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations. @@ -507,6 +530,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info @@ -1098,13 +1127,14 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int reg[SOC_ENUM_MAX_REGS]; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int mask[SOC_ENUM_MAX_REGS]; const char * const *texts; const unsigned int *values; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..e47479d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,19 +2636,125 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** + * snd_soc_info_enum_wide - enumerated mulitple register mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a enumerated multiple register + * mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = e->items; + + if (uinfo->value.enumerated.item > e->items - 1) + uinfo->value.enumerated.item = e->items - 1; + strcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item]); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_enum_wide); + +/** + * snd_soc_get_value_enum_wide - semi enumerated multiple registers mixer + * get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a semi enumerated mutiple registers mixer. + * + * Mutiple semi enumerated mixer: the mixer has multiple registers to set the + * values. The enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, i, reg_idx; + bool match = true; + + for (i = 0; i < e->items; i++) { + match = true; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + val = snd_soc_read(codec, e->reg[reg_idx]); + if (val != e->values[i * e->num_regs + reg_idx]) { + match = false; + break; + } + } + if (match) + break; + } + if (!match) { + dev_err(codec->dev, "ASoC: Failed to find matched enum value\n"); + return -EINVAL; + } else + ucontrol->value.enumerated.item[0] = i; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_wide); + +/** + * snd_soc_get_value_enum_wide - semi enumerated multiple mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a multiple semi enumerated mixer. + * + * Mutiple semi enumerated mixer: the mixer has multiple registers to set the + * values. The enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, reg_idx, item; + int ret; + + if (ucontrol->value.enumerated.item[0] > e->items - 1) + return -EINVAL; + item = ucontrol->value.enumerated.item[0]; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + val = e->values[item * e->num_regs + reg_idx]; + ret = snd_soc_update_bits_locked(codec, e->reg[reg_idx], + e->mask[reg_idx], val); + if (ret) + return ret; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_wide); + +/** * snd_soc_read_signed - Read a codec register and interprete as signed value * @codec: codec * @reg: Register to read diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..22ca178 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, unsigned int val, item; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; + if (e->reg[0] != SND_SOC_NOPM) { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); } else { /* since a virtual mux has no backing registers to @@ -1553,7 +1553,7 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; + unsigned int wi, update_idx; int ret; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) @@ -1575,8 +1575,10 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); + for (update_idx = 0; update_idx < update->num_regs; update_idx++) + ret = soc_widget_update_bits_locked(w, update->reg[update_idx], + update->mask[update_idx], + update->val[update_idx]); if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2866,9 +2868,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2906,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2948,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +2988,115 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_value_enum_wide - dapm semi enumerated multiple registers + * mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple regsters to set the enumerated items. The enumerated + * iteams are referred as values. + * Can be used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, i, reg_idx; + bool match = true; + + for (i = 0; i < e->items; i++) { + match = true; + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + if (reg_val != e->values[i * e->num_regs + reg_idx]) { + match = false; + break; + } + } + if (match) + break; + } + if (!match) { + dev_err(codec->dev, "ASoC: Failed to find matched enum value\n"); + return -EINVAL; + } else + ucontrol->value.enumerated.item[0] = i; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_wide); + +/** + * snd_soc_dapm_put_value_enum_wide - dapm semi enumerated multiple registers + * mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple regsters to set the enumerated items. The enumerated + * iteams are referred as values. + * Can be used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_value_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int value, item, old, reg_idx; + struct snd_soc_dapm_update update; + int wi, update_idx; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] > e->items - 1) + return -EINVAL; + + item = ucontrol->value.enumerated.item[0]; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + for (reg_idx = 0, update_idx = 0; reg_idx < e->num_regs; reg_idx++) { + value = e->values[item * e->num_regs + reg_idx]; + old = snd_soc_read(codec, e->reg[reg_idx]); + if (value != old) { + update.reg[update_idx] = e->reg[reg_idx]; + update.mask[update_idx] = e->mask[reg_idx]; + update.val[update_idx] = value; + update.num_regs = ++update_idx; + } + } + + if (update_idx) { + update.kcontrol = kcontrol; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item, e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_wide); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 7

[alsa-devel] [PATCH] ASoC: rt5640: Correct the judgement of data length
by Oder Chiou 27 Mar '14

27 Mar '14

The patch corrects the judgement of data length. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0883c8a..bcd6513 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1622,16 +1622,16 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", bclk_ms, pre_div, dai->id); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val_len |= RT5640_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val_len |= RT5640_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: val_len |= RT5640_I2S_DL_8; break; default: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Add isr to deal with error flag
by Nicolin Chen 27 Mar '14

27 Mar '14

It's quite cricial to clear error flags because SAI might hang if getting FIFO underrun during playback (I haven't confirmed the same issue on Rx overflow though). So this patch enables those irq and adds isr() to clear the flags so as to keep playback entirely safe. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 9 +++++++ 2 files changed, 75 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c4a4231..5f91aff 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,55 @@ #include "fsl_sai.h" +#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ + FSL_SAI_CSR_FEIE |\ + FSL_SAI_CSR_FWIE) + +static irqreturn_t fsl_sai_isr(int irq, void *devid) +{ + struct fsl_sai *sai = (struct fsl_sai *)devid; + struct device *dev = &sai->pdev->dev; + u32 xcsr; + + regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, xcsr); + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Tx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_dbg(dev, "isr: Tx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) + dev_dbg(dev, "isr: Transmit underrun detected\n"); + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n"); + + regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, xcsr); + + if (xcsr & FSL_SAI_CSR_WSF) + dev_dbg(dev, "isr: Start of Rx word detected\n"); + + if (xcsr & FSL_SAI_CSR_SEF) + dev_dbg(dev, "isr: Rx Frame sync error detected\n"); + + if (xcsr & FSL_SAI_CSR_FEF) + dev_dbg(dev, "isr: Receive overflow detected\n"); + + if (xcsr & FSL_SAI_CSR_FWF) + dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); + + if (xcsr & FSL_SAI_CSR_FRF) + dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n"); + + return IRQ_HANDLED; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -373,8 +422,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -490,12 +539,14 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int ret; + int irq, ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->pdev = pdev; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -514,6 +565,18 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(sai->regmap); } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e432260..05d1a1f 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -37,7 +37,15 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_WSF BIT(20) +#define FSL_SAI_CSR_SEF BIT(19) +#define FSL_SAI_CSR_FEF BIT(18) #define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRF BIT(16) +#define FSL_SAI_CSR_WSIE BIT(12) +#define FSL_SAI_CSR_SEIE BIT(11) +#define FSL_SAI_CSR_FEIE BIT(10) +#define FSL_SAI_CSR_FWIE BIT(9) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) @@ -99,6 +107,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { + struct platform_device *pdev; struct regmap *regmap; bool big_endian_regs; -- 1.8.4

4 15

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 27 Mar '14

27 Mar '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Move cache sync() to resume()
by Oder Chiou 27 Mar '14

27 Mar '14

The patch fixes the defect in case of resume which doesn't sync the cache. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0061ae6..0883c8a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1890,11 +1890,9 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { - regcache_cache_only(rt5640->regmap, false); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2, @@ -1904,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); - regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); snd_soc_update_bits(codec, RT5640_MICBIAS, @@ -1979,6 +1976,9 @@ static int rt5640_resume(struct snd_soc_codec *codec) msleep(400); } + regcache_cache_only(rt5640->regmap, false); + regcache_sync(rt5640->regmap); + return 0; } #else -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Thanks to Peter for the review and bearing with my confusion :) sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 375e197..1fbded0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + u8 *bytes = (u8*) &val; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, val); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: set up user bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 55 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 53 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 712a7cd..ec0463a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -566,8 +567,11 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -589,6 +593,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 1, + IEC958_AES1_CON_PCM_CODER); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 1, + IEC958_AES1_CON_PCM_CODER); + + switch (rate) { + case 22050: + val |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + val |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + val |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + val |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + val |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + val |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 3, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 3, val); + return 0; } @@ -621,7 +672,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

3 5

[alsa-devel] Lenovo G580 mic noise when recording
by Nevyan Neykov 27 Mar '14

27 Mar '14

Hello, my lenovo g580(2189, 20150) laptop is using intel hd audio conexant driver( Intel Corporation 7 Series/C210 ) and experiencing lots of noise when trying to record from either external or internal microphone. If I lower mic volume / boost the voice get barely recorded. I am using this line in my alsa conf: options snd-hda-intel model=auto position_fix=2 also tried with model=asus etc..., when using position_fix=1 recording receives lots of cracking noise. I must mention that both mics are working very well under win7/8. Regards.

1 0

[alsa-devel] [PATCH RFC v3 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 27 Mar '14

27 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. These patches implement the main part of the simple-card changes discussed in alsa-devel mailing list [2]. Since RFC v2 version of the patches, address remaining comments from Jean-Francois Moine [4]: - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Only allow legacy parsing if there is no dai-link node. - Fix double unref in of_card parsing loop. Since RFC version of the patches: - ASoC: core: Update snd_soc_of_parse_daifmt() interface - Make minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). (Kuninori Morimoto's comment [3]). - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Fix typo in DT binding document (Jean-Francois Moine's comment [4]). - Remove #define DEBUG - Remove accidental line wrap at simple-audio-card,name property descripition in the DT binding document [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074388.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074699.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074697.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 88 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 238 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 200 insertions(+), 138 deletions(-) -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: jack: export gpio detect
by cfreeman＠nvidia.com 26 Mar '14

26 Mar '14

From: Christopher Freeman <cfreeman(a)nvidia.com> Export the gpio detect function so machine drivers may call it. Interrupts for the jack may be disabled during sleep, so this allows a machine driver to have the jack status updated during resume. Signed-off-by: Christopher Freeman <cfreeman(a)nvidia.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..36383cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -442,6 +442,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio); #endif /* codec register bit access */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..fd97cc5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -231,7 +231,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); #ifdef CONFIG_GPIOLIB /* gpio detect */ -static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; @@ -251,6 +251,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) snd_soc_jack_report(jack, report, gpio->report); } +EXPORT_SYMBOL_GPL(snd_soc_jack_gpio_detect); /* irq handler for gpio pin */ static irqreturn_t gpio_handler(int irq, void *data) -- 1.8.3.2 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

1 0

Re: [alsa-devel] linux-next: Tree for Mar 26 (sound/soc/codecs/cs42l52.c)
by Randy Dunlap 26 Mar '14

26 Mar '14

On 03/25/2014 11:35 PM, Stephen Rothwell wrote: > Hi all, > > This tree still fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140325: > on x86_64: when CONFIG_INPUT=m and SNC_SOC_CS42L42=y: sound/built-in.o: In function `cs42l52_beep_set': cs42l52.c:(.text+0xd4b01): undefined reference to `input_event' sound/built-in.o: In function `cs42l52_probe': cs42l52.c:(.text+0xd4beb): undefined reference to `devm_input_allocate_device' cs42l52.c:(.text+0xd4d0c): undefined reference to `input_register_device' sound/built-in.o: In function `wm8962_beep_set': wm8962.c:(.text+0x116240): undefined reference to `input_event' sound/built-in.o: In function `wm8962_probe': wm8962.c:(.text+0x116cae): undefined reference to `devm_input_allocate_device' wm8962.c:(.text+0x116e0e): undefined reference to `input_register_device' Full randconfig file is attached. -- ~Randy

1 1

[alsa-devel] Fwd: Re: no reset_resume for driver snd-usb-audio for logitech headset H600
by baumber＠hotmail.com 26 Mar '14

26 Mar '14

FWD: Any news to the snd-usb-audio issue? Thanks, Bernhard Old Message: ------------- I have tested the patch with kernel 3.12.0 with suspend and hibernate, and the problem is gone. No messages "no reset_resume for driver snd-usb-audio?" or other negative effects/messages. After resume, the usb_audio_device (Headset) is the master channel in KDE Mixer, as it should be. => So the problem is solved for me. Can you please add the patch to the latest kernels and LTS-kernels or whitelist the workaround for my device (ID 046d:0a29 Logitech, Inc. H600 [Wireless Headset]) Thanks, Bernhard On 2013-10-20 23:35, Takashi Iwai wrote: > At Sun, 20 Oct 2013 23:18:13 +0200, > <baumber(a)hotmail.com> wrote: >> >> On 2013-10-09 11:16 (CEST,UTC+2), Takashi Iwai wrote: >>>>>> diff --git a/sound/usb/card.c b/sound/usb/card.c >>>>>> index 9d9de8d..c591a83 100644 >>>>>> --- a/sound/usb/card.c >>>>>> +++ b/sound/usb/card.c >>>>>> @@ -747,6 +747,7 @@ static struct usb_driver usb_audio_driver = { >>>>>> .disconnect = usb_audio_disconnect, >>>>>> .suspend = usb_audio_suspend, >>>>>> .resume = usb_audio_resume, >>>>>> + .reset_resume = usb_audio_resume, >>>>>> .id_table = usb_audio_ids, >>>>>> .supports_autosuspend = 1, >>>>>> }; >> >> Shall I add the line in b/sound/usb/card.c and try it? Which kernel version shall I use? > > It doesn't matter which kernel version is. > The error message should disappear, but I don't know whether any other > positive effect you get. As Clemens pointed, there can be some > negative effects by this, so it cannot be taken as is. But at least, > it'd be good to know the result... > > > thanks, > > Takashi

3 10

[alsa-devel] [PATCH 1/3] ASoC: ak5386: add regulator consumer support
by Daniel Mack 26 Mar '14

26 Mar '14

The chip has two power supplies, VA and VDD. Register and enable them both. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak5386.c | 35 ++++++++++++++++++++++++++++++++--- 1 file changed, 32 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b..99fb5db 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -122,6 +128,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +137,19 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); @@ -140,13 +160,22 @@ static int ak5386_probe(struct platform_device *pdev) "AK5386 Reset")) priv->reset_gpio = -EINVAL; - return snd_soc_register_codec(dev, &soc_codec_ak5386, - &ak5386_dai, 1); + ret = snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); + if (ret < 0) + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + return ret; } static int ak5386_remove(struct platform_device *pdev) { - snd_soc_unregister_codec(&pdev->dev); + struct device *dev = &pdev->dev; + struct ak5386_priv *priv = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } -- 1.8.5.3

2 5

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 26 Mar '14

26 Mar '14

I've previously posted under ALSA's user mailing list so to avoid the clutter, the details about what's happening is here: http://article.gmane.org/gmane.linux.alsa.user/38258 http://article.gmane.org/gmane.linux.alsa.user/38260 There was no debug info under /proc/asound/card0/ so nothing to give from there unless you guys need something specific. Output of alsa-debug.sh: http://www.alsa-project.org/db/?f=26187faadd47f3626029ba667cb004e74695f2b2 aplay returns: ________________________________________________________________________ $ aplay -vv 03\ paris\ match-\ Bikeride.wav Playing WAVE '03 paris match- Bikeride.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 4096 stop_threshold : 4096 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'C-Media CMI8738' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 23219 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 #+ | 00% _____________________________________________________________________

1 0

[alsa-devel] twl4030 latency update
by Leonardo Gabrielli 26 Mar '14

26 Mar '14

Dear Peter, I was investigating on TWL4030 high playback latency and stumbled in an old thread started by Edgar http://mailman.alsa-project.org/pipermail/alsa-devel/2011-October/045173.ht… where I read this is related to McBSP2 buffer length Recent kernels seems to have the same behavior (I have a debian beagleboardxM with 3.13.3-armv7-x10) Did you manage to get a fix to this problem? Would it be possible? Regards Leonardo -- Dr. Leonardo Gabrielli, PhD student A3Lab - Dept. Information Engineering Università Politecnica delle Marche via Brecce Bianche 12, 60131, Ancona, Italy Skype: leonardo.gabrielli Web: a3lab.dii.univpm.it/people/leonardo-gabrielli <http://a3lab.dii.univpm.it/people/leonardo-gabrielli>

2 11

[alsa-devel] [PATCH] ASoC: add RT5639 CODEC driver
by oder_chiou＠realtek.com 26 Mar '14

26 Mar '14

From: Oder Chiou <oder_chiou(a)realtek.com> This patch adds the RT5639 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- It is migrated from RT5640 codec driver, and modified for RT5639's specification. --- include/sound/rt5639.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5639.c | 2092 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5639.h | 1928 +++++++++++++++++++++++++++++++++++++++++ 5 files changed, 4048 insertions(+) create mode 100644 include/sound/rt5639.h create mode 100644 sound/soc/codecs/rt5639.c create mode 100644 sound/soc/codecs/rt5639.h diff --git a/include/sound/rt5639.h b/include/sound/rt5639.h new file mode 100644 index 0000000..7101120 --- /dev/null +++ b/include/sound/rt5639.h @@ -0,0 +1,22 @@ +/* + * linux/sound/rt5639.h -- Platform data for RT5639 + * + * Copyright 2011 Realtek Microelectronics + * + * 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 __LINUX_SND_RT5639_H +#define __LINUX_SND_RT5639_H + +struct rt5639_platform_data { + /* IN1 & IN2 can optionally be differential */ + bool in1_diff; + bool in2_diff; + + int ldo1_en; /* GPIO for LDO1_EN */ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..ed791fb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C + select SND_SOC_RT5639 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -316,6 +317,9 @@ config SND_SOC_PCM3008 config SND_SOC_RT5631 tristate +config SND_SOC_RT5639 + tristate + config SND_SOC_RT5640 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..051f3fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o +snd-soc-rt5639-objs := rt5639.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -180,6 +181,7 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o +obj-$(CONFIG_SND_SOC_RT5639) += snd-soc-rt5639.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5639.c b/sound/soc/codecs/rt5639.c new file mode 100644 index 0000000..df8d307 --- /dev/null +++ b/sound/soc/codecs/rt5639.c @@ -0,0 +1,2092 @@ +/* + * rt5639.c -- RT5639 ALSA SoC audio codec driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * 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/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.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 <sound/tlv.h> + +#include "rt5639.h" + +#define RT5639_DEVICE_ID 0x6231 + +#define RT5639_PR_RANGE_BASE (0xff + 1) +#define RT5639_PR_SPACING 0x100 + +#define RT5639_PR_BASE (RT5639_PR_RANGE_BASE + (0 * RT5639_PR_SPACING)) + +static const struct regmap_range_cfg rt5639_ranges[] = { + { .name = "PR", .range_min = RT5639_PR_BASE, + .range_max = RT5639_PR_BASE + 0xb4, + .selector_reg = RT5639_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5639_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + {0x2e, 0x0c00}, + {RT5639_PR_BASE + 0x3d, 0x3600}, + {RT5639_PR_BASE + 0x12, 0x0aa8}, + {RT5639_PR_BASE + 0x14, 0x0aaa}, + {RT5639_PR_BASE + 0x20, 0x6110}, + {RT5639_PR_BASE + 0x21, 0xe0e0}, + {RT5639_PR_BASE + 0x23, 0x1804}, +}; +#define RT5639_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5639_reg[] = { + { 0x00, 0x000e }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x04, 0x8000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0000 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5454 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaa00 }, + { 0x2d, 0x0000 }, + { 0x2e, 0xa000 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf800 }, + { 0x49, 0x3800 }, + { 0x4a, 0x0004 }, + { 0x4c, 0xfc00 }, + { 0x4d, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1114 }, + { 0x74, 0x0c00 }, + { 0x75, 0x1d00 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0008 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0600 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c00 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb8, 0x034b }, + { 0xb9, 0x0066 }, + { 0xba, 0x000b }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0400 }, + { 0xc2, 0x0000 }, + { 0xc4, 0x0000 }, + { 0xc5, 0x0000 }, + { 0xc6, 0x2000 }, + { 0xc8, 0x0000 }, + { 0xc9, 0x0000 }, + { 0xca, 0x0000 }, + { 0xcb, 0x0000 }, + { 0xcc, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd2, 0x8c00 }, + { 0xd3, 0xaa20 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6231 }, +}; + +static int rt5639_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5639_RESET, 0); +} + +static bool rt5639_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5639_ranges); i++) + if ((reg >= rt5639_ranges[i].window_start && + reg <= rt5639_ranges[i].window_start + + rt5639_ranges[i].window_len) || + (reg >= rt5639_ranges[i].range_min && + reg <= rt5639_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5639_RESET: + case RT5639_ASRC_5: + case RT5639_EQ_CTRL1: + case RT5639_DRC_AGC_1: + case RT5639_IRQ_CTRL2: + case RT5639_INT_IRQ_ST: + case RT5639_PRIV_INDEX: + case RT5639_PRIV_DATA: + case RT5639_PGM_REG_ARR1: + case RT5639_PGM_REG_ARR3: + case RT5639_VENDOR_ID: + case RT5639_VENDOR_ID1: + case RT5639_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5639_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5639_ranges); i++) + if ((reg >= rt5639_ranges[i].window_start && + reg <= rt5639_ranges[i].window_start + + rt5639_ranges[i].window_len) || + (reg >= rt5639_ranges[i].range_min && + reg <= rt5639_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5639_RESET: + case RT5639_SPK_VOL: + case RT5639_HP_VOL: + case RT5639_OUTPUT: + case RT5639_IN1_IN2: + case RT5639_IN3_IN4: + case RT5639_INL_INR_VOL: + case RT5639_DAC1_DIG_VOL: + case RT5639_DAC2_DIG_VOL: + case RT5639_DAC2_CTRL: + case RT5639_ADC_DIG_VOL: + case RT5639_ADC_DATA: + case RT5639_ADC_BST_VOL: + case RT5639_STO_ADC_MIXER: + case RT5639_MONO_ADC_MIXER: + case RT5639_AD_DA_MIXER: + case RT5639_STO_DAC_MIXER: + case RT5639_MONO_DAC_MIXER: + case RT5639_DIG_MIXER: + case RT5639_DIG_INF_DATA: + case RT5639_REC_L1_MIXER: + case RT5639_REC_L2_MIXER: + case RT5639_REC_R1_MIXER: + case RT5639_REC_R2_MIXER: + case RT5639_HPO_MIXER: + case RT5639_SPK_L_MIXER: + case RT5639_SPK_R_MIXER: + case RT5639_SPO_L_MIXER: + case RT5639_SPO_R_MIXER: + case RT5639_SPO_CLSD_RATIO: + case RT5639_OUT_L1_MIXER: + case RT5639_OUT_L2_MIXER: + case RT5639_OUT_L3_MIXER: + case RT5639_OUT_R1_MIXER: + case RT5639_OUT_R2_MIXER: + case RT5639_OUT_R3_MIXER: + case RT5639_LOUT_MIXER: + case RT5639_PWR_DIG1: + case RT5639_PWR_DIG2: + case RT5639_PWR_ANLG1: + case RT5639_PWR_ANLG2: + case RT5639_PWR_MIXER: + case RT5639_PWR_VOL: + case RT5639_PRIV_INDEX: + case RT5639_PRIV_DATA: + case RT5639_I2S1_SDP: + case RT5639_I2S2_SDP: + case RT5639_ADDA_CLK1: + case RT5639_DMIC: + case RT5639_GLB_CLK: + case RT5639_PLL_CTRL1: + case RT5639_PLL_CTRL2: + case RT5639_ASRC_1: + case RT5639_ASRC_2: + case RT5639_ASRC_3: + case RT5639_ASRC_4: + case RT5639_ASRC_5: + case RT5639_CLS_D_OVCD: + case RT5639_CLS_D_OUT: + case RT5639_DEPOP_M1: + case RT5639_DEPOP_M2: + case RT5639_DEPOP_M3: + case RT5639_CHARGE_PUMP: + case RT5639_PV_DET_SPK_G: + case RT5639_MICBIAS: + case RT5639_EQ_CTRL1: + case RT5639_EQ_CTRL2: + case RT5639_WIND_FILTER: + case RT5639_DRC_AGC_1: + case RT5639_DRC_AGC_2: + case RT5639_DRC_AGC_3: + case RT5639_JD_CTRL: + case RT5639_IRQ_CTRL1: + case RT5639_IRQ_CTRL2: + case RT5639_INT_IRQ_ST: + case RT5639_GPIO_CTRL1: + case RT5639_GPIO_CTRL2: + case RT5639_PGM_REG_ARR1: + case RT5639_PGM_REG_ARR2: + case RT5639_PGM_REG_ARR3: + case RT5639_PGM_REG_ARR4: + case RT5639_PGM_REG_ARR5: + case RT5639_ADJ_HPF: + case RT5639_HP_CALIB_AMP_DET: + case RT5639_SV_ZCD1: + case RT5639_DUMMY1: + case RT5639_DUMMY2: + case RT5639_DUMMY3: + case RT5639_VENDOR_ID: + case RT5639_VENDOR_ID1: + case RT5639_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5639_data_select[] = { + "Normal", "left copy to right", "right copy to left", "Swap"}; + +static SOC_ENUM_SINGLE_DECL(rt5639_if1_dac_enum, RT5639_DIG_INF_DATA, + RT5639_IF1_DAC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if1_adc_enum, RT5639_DIG_INF_DATA, + RT5639_IF1_ADC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if2_dac_enum, RT5639_DIG_INF_DATA, + RT5639_IF2_DAC_SEL_SFT, rt5639_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5639_if2_adc_enum, RT5639_DIG_INF_DATA, + RT5639_IF2_ADC_SEL_SFT, rt5639_data_select); + +/* Class D speaker gain ratio */ +static const char * const rt5639_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", + "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; + +static SOC_ENUM_SINGLE_DECL(rt5639_clsd_spk_ratio_enum, + RT5639_CLS_D_OUT, RT5639_CLSD_RATIO_SFT, rt5639_clsd_spk_ratio); + +static const struct snd_kcontrol_new rt5639_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Channel Switch", RT5639_SPK_VOL, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5639_SPK_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* Headphone Output Volume */ + SOC_DOUBLE("HP Channel Switch", RT5639_HP_VOL, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5639_HP_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5639_OUTPUT, + RT5639_L_MUTE_SFT, RT5639_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5639_OUTPUT, + RT5639_VOL_L_SFT, RT5639_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5639_OUTPUT, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5639_DAC2_CTRL, + RT5639_M_DAC_L2_VOL_SFT, RT5639_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5639_DAC1_DIG_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5639_IN1_IN2, RT5639_BST_SFT1, 8, 0, + bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5639_IN3_IN4, RT5639_BST_SFT2, 8, 0, + bst_tlv), + + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5639_INL_INR_VOL, + RT5639_INL_VOL_SFT, RT5639_INR_VOL_SFT, 31, 1, in_vol_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5639_ADC_DIG_VOL, + RT5639_L_MUTE_SFT, RT5639_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5639_ADC_DIG_VOL, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 127, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5639_ADC_DATA, + RT5639_L_VOL_SFT, RT5639_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5639_ADC_BST_VOL, + RT5639_ADC_L_BST_SFT, RT5639_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + /* Class D speaker gain ratio */ + SOC_ENUM("Class D SPK Ratio Control", rt5639_clsd_spk_ratio_enum), + + /* Data Swap */ + SOC_ENUM("ADC IF1 Data Switch", rt5639_if1_adc_enum), + SOC_ENUM("DAC IF1 Data Switch", rt5639_if1_dac_enum), + SOC_ENUM("ADC IF2 Data Switch", rt5639_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5639_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int i, idx = -EINVAL; + int rate, red, bound, temp; + + rate = rt5639->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5639_DMIC, RT5639_DMIC_CLK_MASK, + idx << RT5639_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5639_GLB_CLK); + val &= RT5639_SCLK_SRC_MASK; + if (val == RT5639_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5639_sto_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_STO_ADC_MIXER, + RT5639_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5639_MONO_ADC_MIXER, + RT5639_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5639_AD_DA_MIXER, + RT5639_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5639_AD_DA_MIXER, + RT5639_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5639_AD_DA_MIXER, + RT5639_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5639_AD_DA_MIXER, + RT5639_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_STO_DAC_MIXER, + RT5639_M_DAC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_MONO_DAC_MIXER, + RT5639_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dig_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_DIG_MIXER, + RT5639_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_DIG_MIXER, + RT5639_M_DAC_L2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_dig_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_DIG_MIXER, + RT5639_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_DIG_MIXER, + RT5639_M_DAC_R2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5639_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5639_REC_L2_MIXER, + RT5639_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_REC_L2_MIXER, + RT5639_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST4_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_REC_L2_MIXER, + RT5639_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5639_REC_L2_MIXER, + RT5639_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5639_REC_R2_MIXER, + RT5639_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_REC_R2_MIXER, + RT5639_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST4_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_REC_R2_MIXER, + RT5639_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5639_REC_R2_MIXER, + RT5639_M_OM_R_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ +static const struct snd_kcontrol_new rt5639_spk_l_mix[] = { + SOC_DAPM_SINGLE("REC MIXL Switch", RT5639_SPK_L_MIXER, + RT5639_M_RM_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_SPK_L_MIXER, + RT5639_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_SPK_L_MIXER, + RT5639_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_SPK_L_MIXER, + RT5639_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5639_SPK_L_MIXER, + RT5639_M_OM_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spk_r_mix[] = { + SOC_DAPM_SINGLE("REC MIXR Switch", RT5639_SPK_R_MIXER, + RT5639_M_RM_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_SPK_R_MIXER, + RT5639_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPK_R_MIXER, + RT5639_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_SPK_R_MIXER, + RT5639_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5639_SPK_R_MIXER, + RT5639_M_OM_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5639_OUT_L3_MIXER, + RT5639_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5639_OUT_L3_MIXER, + RT5639_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_R2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_OUT_L3_MIXER, + RT5639_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5639_OUT_R3_MIXER, + RT5639_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5639_OUT_R3_MIXER, + RT5639_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_L2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_OUT_R3_MIXER, + RT5639_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5639_SPO_L_MIXER, + RT5639_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5639_SPO_L_MIXER, + RT5639_M_SV_L_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_SPO_L_MIXER, + RT5639_M_BST1_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_SPO_R_MIXER, + RT5639_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5639_SPO_R_MIXER, + RT5639_M_SV_R_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5639_SPO_R_MIXER, + RT5639_M_BST1_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5639_HPO_MIXER, + RT5639_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5639_HPO_MIXER, + RT5639_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5639_LOUT_MIXER, + RT5639_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5639_LOUT_MIXER, + RT5639_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5639_LOUT_MIXER, + RT5639_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5639_LOUT_MIXER, + RT5639_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_SPK_VOL, + RT5639_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_SPK_VOL, + RT5639_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_HP_VOL, + RT5639_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5639_HP_VOL, + RT5639_R_MUTE_SFT, 1, 1); + +/* Stereo ADC source */ +static const char * const rt5639_stereo_adc1_src[] = { + "DIG MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_stereo_adc1_enum, RT5639_STO_ADC_MIXER, + RT5639_ADC_1_SRC_SFT, rt5639_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5639_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5639_stereo_adc1_enum); + +static const char * const rt5639_stereo_adc2_src[] = { + "DMIC1", "DMIC2", "DIG MIX" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_stereo_adc2_enum, RT5639_STO_ADC_MIXER, + RT5639_ADC_2_SRC_SFT, rt5639_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5639_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5639_stereo_adc2_enum); + +/* Mono ADC source */ +static const char * const rt5639_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADCL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_l1_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_L1_SRC_SFT, rt5639_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5639_mono_adc_l1_enum); + +static const char * const rt5639_mono_adc_l2_src[] = { + "DMIC L1", "DMIC L2", "Mono DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_l2_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_L2_SRC_SFT, rt5639_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5639_mono_adc_l2_enum); + +static const char * const rt5639_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADCR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_r1_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_R1_SRC_SFT, rt5639_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5639_mono_adc_r1_enum); + +static const char * const rt5639_mono_adc_r2_src[] = { + "DMIC R1", "DMIC R2", "Mono DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_mono_adc_r2_enum, RT5639_MONO_ADC_MIXER, + RT5639_MONO_ADC_R2_SRC_SFT, rt5639_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5639_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5639_mono_adc_r2_enum); + +/* digital interface and iis interface map */ +static const char * const rt5639_dai_iis_map[] = { + "1:1|2:2", "1:2|2:1", "1:1|2:1", "1:2|2:2" +}; + +static int rt5639_dai_iis_map_values[] = { + 0, + 5, + 6, + 7, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5639_dai_iis_map_enum, + RT5639_I2S1_SDP, RT5639_I2S_IF_SFT, + 0x7, rt5639_dai_iis_map, + rt5639_dai_iis_map_values); + +static const struct snd_kcontrol_new rt5639_dai_mux = + SOC_DAPM_VALUE_ENUM("DAI select", rt5639_dai_iis_map_enum); + +/* SDI select */ +static const char * const rt5639_sdi_sel[] = { + "IF1", "IF2" +}; + +static SOC_ENUM_SINGLE_DECL(rt5639_sdi_sel_enum, RT5639_I2S2_SDP, + RT5639_I2S2_SDI_SFT, rt5639_sdi_sel); + +static const struct snd_kcontrol_new rt5639_sdi_mux = + SOC_DAPM_ENUM("SDI select", rt5639_sdi_sel_enum); + +static int rt5639_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5639_GPIO_CTRL1, + RT5639_GP2_PIN_MASK | RT5639_GP3_PIN_MASK, + RT5639_GP2_PIN_DMIC1_SCL | RT5639_GP3_PIN_DMIC1_SDA); + snd_soc_update_bits(codec, RT5639_DMIC, + RT5639_DMIC_1L_LH_MASK | RT5639_DMIC_1R_LH_MASK | + RT5639_DMIC_1_DP_MASK, + RT5639_DMIC_1L_LH_FALLING | RT5639_DMIC_1R_LH_RISING | + RT5639_DMIC_1_DP_IN1P); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5639_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5639_GPIO_CTRL1, + RT5639_GP2_PIN_MASK | RT5639_GP4_PIN_MASK, + RT5639_GP2_PIN_DMIC1_SCL | RT5639_GP4_PIN_DMIC2_SDA); + snd_soc_update_bits(codec, RT5639_DMIC, + RT5639_DMIC_2L_LH_MASK | RT5639_DMIC_2R_LH_MASK | + RT5639_DMIC_2_DP_MASK, + RT5639_DMIC_2L_LH_FALLING | RT5639_DMIC_2R_LH_RISING | + RT5639_DMIC_2_DP_IN1N); + break; + + default: + return 0; + } + + return 0; +} + +static void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5639->regmap, RT5639_PR_BASE + + RT5639_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M2, + RT5639_DEPOP_MASK, RT5639_DEPOP_MAN); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M1, + RT5639_HP_CP_MASK | RT5639_HP_SG_MASK | RT5639_HP_CB_MASK, + RT5639_HP_CP_PU | RT5639_HP_SG_DIS | RT5639_HP_CB_PU); + regmap_write(rt5639->regmap, RT5639_PR_BASE + RT5639_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2, 0); + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_HA, + RT5639_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5639->regmap, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2 , + RT5639_PWR_FV1 | RT5639_PWR_FV2); +} + +static void rt5639_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M2, + RT5639_DEPOP_MASK | RT5639_DIG_DP_MASK, + RT5639_DEPOP_AUTO | RT5639_DIG_DP_EN); + regmap_update_bits(rt5639->regmap, RT5639_CHARGE_PUMP, + RT5639_PM_HP_MASK, RT5639_PM_HP_HV); + + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M3, + RT5639_CP_FQ1_MASK | RT5639_CP_FQ2_MASK | RT5639_CP_FQ3_MASK, + (RT5639_CP_FQ_192_KHZ << RT5639_CP_FQ1_SFT) | + (RT5639_CP_FQ_12_KHZ << RT5639_CP_FQ2_SFT) | + (RT5639_CP_FQ_192_KHZ << RT5639_CP_FQ3_SFT)); + + regmap_write(rt5639->regmap, RT5639_PR_BASE + + RT5639_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5639->regmap, RT5639_DEPOP_M1, + RT5639_HP_CP_MASK | RT5639_HP_SG_MASK, + RT5639_HP_CP_PD | RT5639_HP_SG_EN); + regmap_update_bits(rt5639->regmap, RT5639_PR_BASE + + RT5639_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5639_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5639_pmu_depop(codec); + rt5639->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5639->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5639_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5639_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5639->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5639_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5639_PWR_ANLG2, RT5639_PWR_PLL_BIT, + 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO2", RT5639_PWR_ANLG1, RT5639_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5639_PWR_ANLG2, RT5639_PWR_MB1_BIT, + 0, NULL, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5639_DMIC, + RT5639_DMIC_1_EN_SFT, 0, rt5639_set_dmic1_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5639_DMIC, + RT5639_DMIC_2_EN_SFT, 0, rt5639_set_dmic2_event, + SND_SOC_DAPM_PRE_PMU), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5639_PWR_ANLG2, + RT5639_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5639_PWR_ANLG2, + RT5639_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5639_PWR_ANLG2, + RT5639_PWR_BST2_BIT, 0, NULL, 0), + + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5639_PWR_VOL, + RT5639_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5639_PWR_VOL, + RT5639_PWR_IN_R_BIT, 0, NULL, 0), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5639_PWR_MIXER, RT5639_PWR_RM_L_BIT, 0, + rt5639_rec_l_mix, ARRAY_SIZE(rt5639_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5639_PWR_MIXER, RT5639_PWR_RM_R_BIT, 0, + rt5639_rec_r_mix, ARRAY_SIZE(rt5639_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, RT5639_PWR_DIG1, RT5639_PWR_ADC_L_BIT, + 0), + SND_SOC_DAPM_ADC("ADC R", NULL, RT5639_PWR_DIG1, RT5639_PWR_ADC_R_BIT, + 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_2_mux), + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_sto_adc_1_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5639_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("Stereo Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_SF_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_adc_l_mix, ARRAY_SIZE(rt5639_sto_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_adc_r_mix, ARRAY_SIZE(rt5639_sto_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("Mono Left Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_mono_adc_l_mix, ARRAY_SIZE(rt5639_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("Mono Right Filter", RT5639_PWR_DIG2, + RT5639_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_mono_adc_r_mix, ARRAY_SIZE(rt5639_mono_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5639_PWR_DIG1, + RT5639_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5639_PWR_DIG1, + RT5639_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("DAI1 RX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI1 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("SDI1 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_sdi_mux), + SND_SOC_DAPM_MUX("DAI2 RX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF1 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("DAI2 IF2 Mux", SND_SOC_NOPM, 0, 0, &rt5639_dai_mux), + SND_SOC_DAPM_MUX("SDI2 TX Mux", SND_SOC_NOPM, 0, 0, &rt5639_sdi_mux), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_dac_l_mix, ARRAY_SIZE(rt5639_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_dac_r_mix, ARRAY_SIZE(rt5639_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_mono_dac_l_mix, ARRAY_SIZE(rt5639_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_mono_dac_r_mix, ARRAY_SIZE(rt5639_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DIG MIXL", SND_SOC_NOPM, 0, 0, + rt5639_dig_l_mix, ARRAY_SIZE(rt5639_dig_l_mix)), + SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, + rt5639_dig_r_mix, ARRAY_SIZE(rt5639_dig_r_mix)), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5639_PWR_DIG1, RT5639_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5639_PWR_DIG1, RT5639_PWR_DAC_R1_BIT, + 0), + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5639_PWR_DIG1, + RT5639_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5639_PWR_DIG1, + RT5639_PWR_DAC_R2_BIT, 0, NULL, 0), + /* SPK/OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5639_PWR_MIXER, RT5639_PWR_SM_L_BIT, + 0, rt5639_spk_l_mix, ARRAY_SIZE(rt5639_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5639_PWR_MIXER, RT5639_PWR_SM_R_BIT, + 0, rt5639_spk_r_mix, ARRAY_SIZE(rt5639_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5639_PWR_MIXER, RT5639_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5639_PWR_MIXER, RT5639_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_PGA("SPKVOL L", RT5639_PWR_VOL, + RT5639_PWR_SV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", RT5639_PWR_VOL, + RT5639_PWR_SV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL L", RT5639_PWR_VOL, + RT5639_PWR_OV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTVOL R", RT5639_PWR_VOL, + RT5639_PWR_OV_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL L", RT5639_PWR_VOL, + RT5639_PWR_HV_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL R", RT5639_PWR_VOL, + RT5639_PWR_HV_R_BIT, 0, NULL, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, + 0, rt5639_spo_l_mix, ARRAY_SIZE(rt5639_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, + 0, rt5639_spo_r_mix, ARRAY_SIZE(rt5639_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5639_PWR_ANLG1, RT5639_PWR_LM_BIT, 0, + rt5639_lout_mix, ARRAY_SIZE(rt5639_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5639_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5639_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5639_PWR_ANLG1, + RT5639_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5639_PWR_ANLG1, + RT5639_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5639_PWR_DIG1, + RT5639_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5639_hp_post_event), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_route rt5639_dapm_routes[] = { + {"IN1P", NULL, "LDO2"}, + {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + {"DMIC L2", NULL, "DMIC2"}, + {"DMIC R2", NULL, "DMIC2"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, + + {"INL VOL", NULL, "IN2P"}, + {"INR VOL", NULL, "IN2N"}, + + {"RECMIXL", "HPOL Switch", "HPOL"}, + {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, + + {"RECMIXR", "HPOR Switch", "HPOR"}, + {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC L2", NULL, "DMIC CLK"}, + {"DMIC L2", NULL, "DMIC2 Power"}, + {"DMIC R2", NULL, "DMIC CLK"}, + {"DMIC R2", NULL, "DMIC2 Power"}, + + {"Stereo ADC L2 Mux", "DMIC1", "DMIC L1"}, + {"Stereo ADC L2 Mux", "DMIC2", "DMIC L2"}, + {"Stereo ADC L2 Mux", "DIG MIX", "DIG MIXL"}, + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DIG MIX", "DIG MIXL"}, + + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DIG MIX", "DIG MIXR"}, + {"Stereo ADC R2 Mux", "DMIC1", "DMIC R1"}, + {"Stereo ADC R2 Mux", "DMIC2", "DMIC R2"}, + {"Stereo ADC R2 Mux", "DIG MIX", "DIG MIXR"}, + + {"Mono ADC L2 Mux", "DMIC L1", "DMIC L1"}, + {"Mono ADC L2 Mux", "DMIC L2", "DMIC L2"}, + {"Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL"}, + {"Mono ADC L1 Mux", "ADCL", "ADC L"}, + + {"Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + {"Mono ADC R1 Mux", "ADCR", "ADC R"}, + {"Mono ADC R2 Mux", "DMIC R1", "DMIC R1"}, + {"Mono ADC R2 Mux", "DMIC R2", "DMIC R2"}, + {"Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR"}, + + {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, + {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, + {"Stereo ADC MIXL", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, + {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, + {"Stereo ADC MIXR", NULL, "Stereo Filter"}, + {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, + {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, + {"Mono ADC MIXL", NULL, "Mono Left Filter"}, + {"Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, + {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, + {"Mono ADC MIXR", NULL, "Mono Right Filter"}, + {"Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll}, + + {"IF2 ADC L", NULL, "Mono ADC MIXL"}, + {"IF2 ADC R", NULL, "Mono ADC MIXR"}, + {"IF1 ADC L", NULL, "Stereo ADC MIXL"}, + {"IF1 ADC R", NULL, "Stereo ADC MIXR"}, + + {"IF1 ADC", NULL, "I2S1"}, + {"IF1 ADC", NULL, "IF1 ADC L"}, + {"IF1 ADC", NULL, "IF1 ADC R"}, + {"IF2 ADC", NULL, "I2S2"}, + {"IF2 ADC", NULL, "IF2 ADC L"}, + {"IF2 ADC", NULL, "IF2 ADC R"}, + + {"DAI1 TX Mux", "1:1|2:2", "IF1 ADC"}, + {"DAI1 TX Mux", "1:2|2:1", "IF2 ADC"}, + {"DAI1 IF1 Mux", "1:1|2:1", "IF1 ADC"}, + {"DAI1 IF2 Mux", "1:1|2:1", "IF2 ADC"}, + {"SDI1 TX Mux", "IF1", "DAI1 IF1 Mux"}, + {"SDI1 TX Mux", "IF2", "DAI1 IF2 Mux"}, + + {"DAI2 TX Mux", "1:2|2:1", "IF1 ADC"}, + {"DAI2 TX Mux", "1:1|2:2", "IF2 ADC"}, + {"DAI2 IF1 Mux", "1:2|2:2", "IF1 ADC"}, + {"DAI2 IF2 Mux", "1:2|2:2", "IF2 ADC"}, + {"SDI2 TX Mux", "IF1", "DAI2 IF1 Mux"}, + {"SDI2 TX Mux", "IF2", "DAI2 IF2 Mux"}, + + {"AIF1TX", NULL, "DAI1 TX Mux"}, + {"AIF1TX", NULL, "SDI1 TX Mux"}, + {"AIF2TX", NULL, "DAI2 TX Mux"}, + {"AIF2TX", NULL, "SDI2 TX Mux"}, + + {"DAI1 RX Mux", "1:1|2:2", "AIF1RX"}, + {"DAI1 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI1 RX Mux", "1:2|2:1", "AIF2RX"}, + {"DAI1 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"DAI2 RX Mux", "1:2|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:1", "AIF1RX"}, + {"DAI2 RX Mux", "1:1|2:2", "AIF2RX"}, + {"DAI2 RX Mux", "1:2|2:2", "AIF2RX"}, + + {"IF1 DAC", NULL, "I2S1"}, + {"IF1 DAC", NULL, "DAI1 RX Mux"}, + {"IF2 DAC", NULL, "I2S2"}, + {"IF2 DAC", NULL, "DAI2 RX Mux"}, + + {"IF1 DAC L", NULL, "IF1 DAC"}, + {"IF1 DAC R", NULL, "IF1 DAC"}, + {"IF2 DAC L", NULL, "IF2 DAC"}, + {"IF2 DAC R", NULL, "IF2 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, + + {"DAC L1", NULL, "Stereo DAC MIXL"}, + {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R1", NULL, "Stereo DAC MIXR"}, + {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, + {"SPK MIXL", "INL Switch", "INL VOL"}, + {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, + {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, + {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, + {"SPK MIXR", "INR Switch", "INR VOL"}, + {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, + {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, + + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "INL Switch", "INL VOL"}, + {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, + {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, + + {"OUT MIXR", "BST2 Switch", "BST2"}, + {"OUT MIXR", "BST1 Switch", "BST1"}, + {"OUT MIXR", "INR Switch", "INR VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"SPKVOL L", NULL, "SPK MIXL"}, + {"SPKVOL R", NULL, "SPK MIXR"}, + {"HPOVOL L", NULL, "OUT MIXL"}, + {"HPOVOL R", NULL, "OUT MIXR"}, + {"OUTVOL L", NULL, "OUT MIXL"}, + {"OUTVOL R", NULL, "OUT MIXR"}, + + {"SPOL MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOL MIX", "DAC L1 Switch", "DAC L1"}, + {"SPOL MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOL MIX", "SPKVOL L Switch", "SPKVOL L"}, + {"SPOL MIX", "BST1 Switch", "BST1"}, + {"SPOR MIX", "DAC R1 Switch", "DAC R1"}, + {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, + {"SPOR MIX", "BST1 Switch", "BST1"}, + + {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, + {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, + {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, + {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, + + {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, + {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, + {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, + + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, + + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, + + {"SPOLP", NULL, "Improve SPK Amp Drv"}, + {"SPOLN", NULL, "Improve SPK Amp Drv"}, + {"SPORP", NULL, "Improve SPK Amp Drv"}, + {"SPORN", NULL, "Improve SPK Amp Drv"}, + + {"HPOL", NULL, "Improve HP Amp Drv"}, + {"HPOR", NULL, "Improve HP Amp Drv"}, + + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, + {"LOUTL", NULL, "LOUT MIX"}, + {"LOUTR", NULL, "LOUT MIX"}, +}; + +static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) +{ + int ret = 0, val; + + if (codec == NULL) + return -EINVAL; + + val = snd_soc_read(codec, RT5639_I2S1_SDP); + val = (val & RT5639_I2S_IF_MASK) >> RT5639_I2S_IF_SFT; + switch (dai_id) { + case RT5639_AIF1: + switch (val) { + case RT5639_IF_123: + case RT5639_IF_132: + ret |= RT5639_U_IF1; + break; + case RT5639_IF_113: + ret |= RT5639_U_IF1; + case RT5639_IF_312: + case RT5639_IF_213: + ret |= RT5639_U_IF2; + break; + } + break; + + case RT5639_AIF2: + switch (val) { + case RT5639_IF_231: + case RT5639_IF_213: + ret |= RT5639_U_IF1; + break; + case RT5639_IF_223: + ret |= RT5639_U_IF1; + case RT5639_IF_123: + case RT5639_IF_321: + ret |= RT5639_U_IF2; + break; + } + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5639_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int dai_sel, pre_div, bclk_ms, frame_size; + + rt5639->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5639->sysclk, rt5639->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5639->lrck[dai->id], dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + rt5639->bclk[dai->id] = rt5639->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5639->bclk[dai->id], rt5639->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5639_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5639_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5639_I2S_DL_8; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + mask_clk = RT5639_I2S_BCLK_MS1_MASK | RT5639_I2S_PD1_MASK; + val_clk = bclk_ms << RT5639_I2S_BCLK_MS1_SFT | + pre_div << RT5639_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5639_I2S1_SDP, + RT5639_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5639_ADDA_CLK1, mask_clk, val_clk); + } + if (dai_sel & RT5639_U_IF2) { + mask_clk = RT5639_I2S_BCLK_MS2_MASK | RT5639_I2S_PD2_MASK; + val_clk = bclk_ms << RT5639_I2S_BCLK_MS2_SFT | + pre_div << RT5639_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5639_I2S2_SDP, + RT5639_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5639_ADDA_CLK1, mask_clk, val_clk); + } + + return 0; +} + +static int rt5639_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + int dai_sel; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5639->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5639_I2S_MS_S; + rt5639->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5639_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5639_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5639_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5639_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + snd_soc_update_bits(codec, RT5639_I2S1_SDP, + RT5639_I2S_MS_MASK | RT5639_I2S_BP_MASK | + RT5639_I2S_DF_MASK, reg_val); + } + if (dai_sel & RT5639_U_IF2) { + snd_soc_update_bits(codec, RT5639_I2S2_SDP, + RT5639_I2S_MS_MASK | RT5639_I2S_BP_MASK | + RT5639_I2S_DF_MASK, reg_val); + } + + return 0; +} + +static int rt5639_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5639->sysclk && clk_id == rt5639->sysclk_src) + return 0; + + switch (clk_id) { + case RT5639_SCLK_S_MCLK: + reg_val |= RT5639_SCLK_SRC_MCLK; + break; + case RT5639_SCLK_S_PLL1: + reg_val |= RT5639_SCLK_SRC_PLL1; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_SCLK_SRC_MASK, reg_val); + rt5639->sysclk = freq; + rt5639->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + return 0; +} + +/** + * rt5639_pll_calc - Calculate PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calculate M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5639_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5639_pll_code *pll_code) +{ + int max_n = RT5639_PLL_N_MAX, max_m = RT5639_PLL_M_MAX; + int n = 0, m = 0, red, n_t, m_t, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5639_PLL_INP_MAX < freq_in || RT5639_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = (freq_in >> 1) + (freq_in >> 2) * n_t; + if (in_t < 0) + continue; + if (in_t == freq_out) { + bypass = true; + n = n_t; + goto code_find; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - freq_out); + if (red < red_t) { + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = 2; + return 0; +} + +static int rt5639_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + struct rt5639_pll_code *pll_code = &rt5639->pll_code; + int ret, dai_sel; + + if (source == rt5639->pll_src && freq_in == rt5639->pll_in && + freq_out == rt5639->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5639->pll_in = 0; + rt5639->pll_out = 0; + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_SCLK_SRC_MASK, RT5639_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5639_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_MCLK); + break; + case RT5639_PLL1_S_BCLK1: + case RT5639_PLL1_S_BCLK2: + dai_sel = get_sdp_info(codec, dai->id); + if (dai_sel < 0) { + dev_err(codec->dev, + "Failed to get sdp info: %d\n", dai_sel); + return -EINVAL; + } + if (dai_sel & RT5639_U_IF1) { + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_BCLK1); + } + if (dai_sel & RT5639_U_IF2) { + snd_soc_update_bits(codec, RT5639_GLB_CLK, + RT5639_PLL1_SRC_MASK, RT5639_PLL1_SRC_BCLK2); + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5639_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=2\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code); + + snd_soc_write(codec, RT5639_PLL_CTRL1, + pll_code->n_code << RT5639_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5639_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5639_PLL_M_SFT | + pll_code->m_bp << RT5639_PLL_M_BP_SFT); + + rt5639->pll_in = freq_in; + rt5639->pll_out = freq_out; + rt5639->pll_src = source; + + return 0; +} + +static int rt5639_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + regcache_cache_only(rt5639->regmap, false); + snd_soc_update_bits(codec, RT5639_PWR_ANLG1, + RT5639_PWR_VREF1 | RT5639_PWR_MB | + RT5639_PWR_BG | RT5639_PWR_VREF2, + RT5639_PWR_VREF1 | RT5639_PWR_MB | + RT5639_PWR_BG | RT5639_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5639_PWR_ANLG1, + RT5639_PWR_FV1 | RT5639_PWR_FV2, + RT5639_PWR_FV1 | RT5639_PWR_FV2); + regcache_sync(rt5639->regmap); + snd_soc_update_bits(codec, RT5639_DUMMY1, + 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5639_MICBIAS, + 0x0030, 0x0030); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5639_DEPOP_M1, 0x0004); + snd_soc_write(codec, RT5639_DEPOP_M2, 0x1100); + snd_soc_update_bits(codec, RT5639_DUMMY1, 0x1, 0); + snd_soc_write(codec, RT5639_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5639_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5639_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5639_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5639_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5639_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5639_probe(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5639->codec = codec; + codec->control_data = rt5639->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + rt5639_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5639_DUMMY1, 0x0301, 0x0301); + snd_soc_update_bits(codec, RT5639_MICBIAS, 0x0030, 0x0030); + + return 0; +} + +static int rt5639_remove(struct snd_soc_codec *codec) +{ + rt5639_reset(codec); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5639_suspend(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + rt5639_set_bias_level(codec, SND_SOC_BIAS_OFF); + rt5639_reset(codec); + regcache_cache_only(rt5639->regmap, true); + regcache_mark_dirty(rt5639->regmap); + if (gpio_is_valid(rt5639->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5639->pdata.ldo1_en, 0); + + return 0; +} + +static int rt5639_resume(struct snd_soc_codec *codec) +{ + struct rt5639_priv *rt5639 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5639->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5639->pdata.ldo1_en, 1); + msleep(400); + } + + return 0; +} +#else +#define rt5639_suspend NULL +#define rt5639_resume NULL +#endif + +#define RT5639_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5639_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5639_aif_dai_ops = { + .hw_params = rt5639_hw_params, + .set_fmt = rt5639_set_dai_fmt, + .set_sysclk = rt5639_set_dai_sysclk, + .set_pll = rt5639_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5639_dai[] = { + { + .name = "rt5639-aif1", + .id = RT5639_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .ops = &rt5639_aif_dai_ops, + }, + { + .name = "rt5639-aif2", + .id = RT5639_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5639_STEREO_RATES, + .formats = RT5639_FORMATS, + }, + .ops = &rt5639_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5639 = { + .probe = rt5639_probe, + .remove = rt5639_remove, + .suspend = rt5639_suspend, + .resume = rt5639_resume, + .set_bias_level = rt5639_set_bias_level, + .idle_bias_off = true, + .controls = rt5639_snd_controls, + .num_controls = ARRAY_SIZE(rt5639_snd_controls), + .dapm_widgets = rt5639_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5639_dapm_widgets), + .dapm_routes = rt5639_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5639_dapm_routes), +}; + +static const struct regmap_config rt5639_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5639_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5639_ranges) * + RT5639_PR_SPACING), + .volatile_reg = rt5639_volatile_register, + .readable_reg = rt5639_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5639_reg, + .num_reg_defaults = ARRAY_SIZE(rt5639_reg), + .ranges = rt5639_ranges, + .num_ranges = ARRAY_SIZE(rt5639_ranges), +}; + +static const struct i2c_device_id rt5639_i2c_id[] = { + { "rt5639", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5639_i2c_id); + +static int rt5639_parse_dt(struct rt5639_priv *rt5639, struct device_node *np) +{ + rt5639->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5639->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5639->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5639->pdata.ldo1_en) && + (rt5639->pdata.ldo1_en != -ENOENT)) + return rt5639->pdata.ldo1_en; + + return 0; +} + +static int rt5639_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5639_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5639_priv *rt5639; + int ret; + unsigned int val; + + rt5639 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5639_priv), + GFP_KERNEL); + if (NULL == rt5639) + return -ENOMEM; + i2c_set_clientdata(i2c, rt5639); + + if (pdata) { + rt5639->pdata = *pdata; + /* + * Translate zero'd out (default) pdata value to an invalid + * GPIO ID. This makes the pdata and DT paths consistent in + * terms of the value left in this field when no GPIO is + * specified, but means we can't actually use GPIO 0. + */ + if (!rt5639->pdata.ldo1_en) + rt5639->pdata.ldo1_en = -EINVAL; + } else if (i2c->dev.of_node) { + ret = rt5639_parse_dt(rt5639, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5639->pdata.ldo1_en = -EINVAL; + + rt5639->regmap = devm_regmap_init_i2c(i2c, &rt5639_regmap); + if (IS_ERR(rt5639->regmap)) { + ret = PTR_ERR(rt5639->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + if (gpio_is_valid(rt5639->pdata.ldo1_en)) { + ret = devm_gpio_request_one(&i2c->dev, rt5639->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, + "RT5639 LDO1_EN"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", + rt5639->pdata.ldo1_en, ret); + return ret; + } + msleep(400); + } + + regmap_read(rt5639->regmap, RT5639_VENDOR_ID2, &val); + if (val != RT5639_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5640/39\n", val); + return -ENODEV; + } + + regmap_write(rt5639->regmap, RT5639_RESET, 0); + + ret = regmap_register_patch(rt5639->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5639->pdata.in1_diff) + regmap_update_bits(rt5639->regmap, RT5639_IN1_IN2, + RT5639_IN_DF1, RT5639_IN_DF1); + + if (rt5639->pdata.in2_diff) + regmap_update_bits(rt5639->regmap, RT5639_IN3_IN4, + RT5639_IN_DF2, RT5639_IN_DF2); + + rt5639->hp_mute = 1; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5639, + rt5639_dai, ARRAY_SIZE(rt5639_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5639_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5639_i2c_driver = { + .driver = { + .name = "rt5639", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5639_acpi_match), + }, + .probe = rt5639_i2c_probe, + .remove = rt5639_i2c_remove, + .id_table = rt5639_i2c_id, +}; +module_i2c_driver(rt5639_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5639 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5639.h b/sound/soc/codecs/rt5639.h new file mode 100644 index 0000000..b9b7f4a --- /dev/null +++ b/sound/soc/codecs/rt5639.h @@ -0,0 +1,1928 @@ +/* + * rt5639.h -- RT5639 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _RT5639_H +#define _RT5639_H + +#include <sound/rt5639.h> + +/* Info */ +#define RT5639_RESET 0x00 +#define RT5639_VENDOR_ID 0xfd +#define RT5639_VENDOR_ID1 0xfe +#define RT5639_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5639_SPK_VOL 0x01 +#define RT5639_HP_VOL 0x02 +#define RT5639_OUTPUT 0x03 +/* I/O - Input */ +#define RT5639_IN1_IN2 0x0d +#define RT5639_IN3_IN4 0x0e +#define RT5639_INL_INR_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5639_DAC1_DIG_VOL 0x19 +#define RT5639_DAC2_DIG_VOL 0x1a +#define RT5639_DAC2_CTRL 0x1b +#define RT5639_ADC_DIG_VOL 0x1c +#define RT5639_ADC_DATA 0x1d +#define RT5639_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5639_STO_ADC_MIXER 0x27 +#define RT5639_MONO_ADC_MIXER 0x28 +#define RT5639_AD_DA_MIXER 0x29 +#define RT5639_STO_DAC_MIXER 0x2a +#define RT5639_MONO_DAC_MIXER 0x2b +#define RT5639_DIG_MIXER 0x2c +#define RT5639_DIG_INF_DATA 0x2f +/* Mixer - ADC */ +#define RT5639_REC_L1_MIXER 0x3b +#define RT5639_REC_L2_MIXER 0x3c +#define RT5639_REC_R1_MIXER 0x3d +#define RT5639_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5639_HPO_MIXER 0x45 +#define RT5639_SPK_L_MIXER 0x46 +#define RT5639_SPK_R_MIXER 0x47 +#define RT5639_SPO_L_MIXER 0x48 +#define RT5639_SPO_R_MIXER 0x49 +#define RT5639_SPO_CLSD_RATIO 0x4a +#define RT5639_OUT_L1_MIXER 0x4d +#define RT5639_OUT_L2_MIXER 0x4e +#define RT5639_OUT_L3_MIXER 0x4f +#define RT5639_OUT_R1_MIXER 0x50 +#define RT5639_OUT_R2_MIXER 0x51 +#define RT5639_OUT_R3_MIXER 0x52 +#define RT5639_LOUT_MIXER 0x53 +/* Power */ +#define RT5639_PWR_DIG1 0x61 +#define RT5639_PWR_DIG2 0x62 +#define RT5639_PWR_ANLG1 0x63 +#define RT5639_PWR_ANLG2 0x64 +#define RT5639_PWR_MIXER 0x65 +#define RT5639_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5639_PRIV_INDEX 0x6a +#define RT5639_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5639_I2S1_SDP 0x70 +#define RT5639_I2S2_SDP 0x71 +#define RT5639_ADDA_CLK1 0x73 +#define RT5639_DMIC 0x75 +/* Function - Analog */ +#define RT5639_GLB_CLK 0x80 +#define RT5639_PLL_CTRL1 0x81 +#define RT5639_PLL_CTRL2 0x82 +#define RT5639_ASRC_1 0x83 +#define RT5639_ASRC_2 0x84 +#define RT5639_ASRC_3 0x85 +#define RT5639_ASRC_4 0x89 +#define RT5639_ASRC_5 0x8a +#define RT5639_CLS_D_OVCD 0x8c +#define RT5639_CLS_D_OUT 0x8d +#define RT5639_DEPOP_M1 0x8e +#define RT5639_DEPOP_M2 0x8f +#define RT5639_DEPOP_M3 0x90 +#define RT5639_CHARGE_PUMP 0x91 +#define RT5639_PV_DET_SPK_G 0x92 +#define RT5639_MICBIAS 0x93 +/* Function - Digital */ +#define RT5639_EQ_CTRL1 0xb0 +#define RT5639_EQ_CTRL2 0xb1 +#define RT5639_WIND_FILTER 0xb2 +#define RT5639_DRC_AGC_1 0xb4 +#define RT5639_DRC_AGC_2 0xb5 +#define RT5639_DRC_AGC_3 0xb6 +#define RT5639_JD_CTRL 0xbb +#define RT5639_IRQ_CTRL1 0xbd +#define RT5639_IRQ_CTRL2 0xbe +#define RT5639_INT_IRQ_ST 0xbf +#define RT5639_GPIO_CTRL1 0xc0 +#define RT5639_GPIO_CTRL2 0xc2 +#define RT5639_PGM_REG_ARR1 0xc8 +#define RT5639_PGM_REG_ARR2 0xc9 +#define RT5639_PGM_REG_ARR3 0xca +#define RT5639_PGM_REG_ARR4 0xcb +#define RT5639_PGM_REG_ARR5 0xcc +#define RT5639_ADJ_HPF 0xd3 +#define RT5639_HP_CALIB_AMP_DET 0xd6 +#define RT5639_SV_ZCD1 0xd9 +/* Dummy Register */ +#define RT5639_DUMMY1 0xfa +#define RT5639_DUMMY2 0xfb +#define RT5639_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5639_CHPUMP_INT_REG1 0x24 +#define RT5639_MAMP_INT_REG2 0x37 +#define RT5639_3D_SPK 0x63 +#define RT5639_WND_1 0x6c +#define RT5639_WND_2 0x6d +#define RT5639_WND_3 0x6e +#define RT5639_WND_4 0x6f +#define RT5639_WND_5 0x70 +#define RT5639_WND_8 0x73 +#define RT5639_DIP_SPK_INF 0x75 +#define RT5639_HP_DCC_INT1 0x77 +#define RT5639_EQ_BW_LOP 0xa0 +#define RT5639_EQ_GN_LOP 0xa1 +#define RT5639_EQ_FC_BP1 0xa2 +#define RT5639_EQ_BW_BP1 0xa3 +#define RT5639_EQ_GN_BP1 0xa4 +#define RT5639_EQ_FC_BP2 0xa5 +#define RT5639_EQ_BW_BP2 0xa6 +#define RT5639_EQ_GN_BP2 0xa7 +#define RT5639_EQ_FC_BP3 0xa8 +#define RT5639_EQ_BW_BP3 0xa9 +#define RT5639_EQ_GN_BP3 0xaa +#define RT5639_EQ_FC_BP4 0xab +#define RT5639_EQ_BW_BP4 0xac +#define RT5639_EQ_GN_BP4 0xad +#define RT5639_EQ_FC_HIP1 0xae +#define RT5639_EQ_GN_HIP1 0xaf +#define RT5639_EQ_FC_HIP2 0xb0 +#define RT5639_EQ_BW_HIP2 0xb1 +#define RT5639_EQ_GN_HIP2 0xb2 +#define RT5639_EQ_PRE_VOL 0xb3 +#define RT5639_EQ_PST_VOL 0xb4 + +/* global definition */ +#define RT5639_L_MUTE (0x1 << 15) +#define RT5639_L_MUTE_SFT 15 +#define RT5639_VOL_L_MUTE (0x1 << 14) +#define RT5639_VOL_L_SFT 14 +#define RT5639_R_MUTE (0x1 << 7) +#define RT5639_R_MUTE_SFT 7 +#define RT5639_VOL_R_MUTE (0x1 << 6) +#define RT5639_VOL_R_SFT 6 +#define RT5639_L_VOL_MASK (0x3f << 8) +#define RT5639_L_VOL_SFT 8 +#define RT5639_R_VOL_MASK (0x3f) +#define RT5639_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5639_BST_SFT1 12 +#define RT5639_BST_SFT2 8 +#define RT5639_IN_DF1 (0x1 << 7) +#define RT5639_IN_SFT1 7 +#define RT5639_IN_DF2 (0x1 << 6) +#define RT5639_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5639_INL_SEL_MASK (0x1 << 15) +#define RT5639_INL_SEL_SFT 15 +#define RT5639_INL_SEL_IN4P (0x0 << 15) +#define RT5639_INL_SEL_MONOP (0x1 << 15) +#define RT5639_INL_VOL_MASK (0x1f << 8) +#define RT5639_INL_VOL_SFT 8 +#define RT5639_INR_SEL_MASK (0x1 << 7) +#define RT5639_INR_SEL_SFT 7 +#define RT5639_INR_SEL_IN4N (0x0 << 7) +#define RT5639_INR_SEL_MONON (0x1 << 7) +#define RT5639_INR_VOL_MASK (0x1f) +#define RT5639_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5639_DAC_L1_VOL_MASK (0xff << 8) +#define RT5639_DAC_L1_VOL_SFT 8 +#define RT5639_DAC_R1_VOL_MASK (0xff) +#define RT5639_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5639_DAC_L2_VOL_MASK (0xff << 8) +#define RT5639_DAC_L2_VOL_SFT 8 +#define RT5639_DAC_R2_VOL_MASK (0xff) +#define RT5639_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5639_M_DAC_L2_VOL (0x1 << 13) +#define RT5639_M_DAC_L2_VOL_SFT 13 +#define RT5639_M_DAC_R2_VOL (0x1 << 12) +#define RT5639_M_DAC_R2_VOL_SFT 12 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5639_ADC_L_VOL_MASK (0x7f << 8) +#define RT5639_ADC_L_VOL_SFT 8 +#define RT5639_ADC_R_VOL_MASK (0x7f) +#define RT5639_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5639_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5639_MONO_ADC_L_VOL_SFT 8 +#define RT5639_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5639_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5639_ADC_L_BST_MASK (0x3 << 14) +#define RT5639_ADC_L_BST_SFT 14 +#define RT5639_ADC_R_BST_MASK (0x3 << 12) +#define RT5639_ADC_R_BST_SFT 12 +#define RT5639_ADC_COMP_MASK (0x3 << 10) +#define RT5639_ADC_COMP_SFT 10 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5639_M_ADC_L1 (0x1 << 14) +#define RT5639_M_ADC_L1_SFT 14 +#define RT5639_M_ADC_L2 (0x1 << 13) +#define RT5639_M_ADC_L2_SFT 13 +#define RT5639_ADC_1_SRC_MASK (0x1 << 12) +#define RT5639_ADC_1_SRC_SFT 12 +#define RT5639_ADC_1_SRC_ADC (0x1 << 12) +#define RT5639_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5639_ADC_2_SRC_MASK (0x3 << 10) +#define RT5639_ADC_2_SRC_SFT 10 +#define RT5639_ADC_2_SRC_DMIC1 (0x0 << 10) +#define RT5639_ADC_2_SRC_DMIC2 (0x1 << 10) +#define RT5639_ADC_2_SRC_DACMIX (0x2 << 10) +#define RT5639_M_ADC_R1 (0x1 << 6) +#define RT5639_M_ADC_R1_SFT 6 +#define RT5639_M_ADC_R2 (0x1 << 5) +#define RT5639_M_ADC_R2_SFT 5 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5639_M_MONO_ADC_L1 (0x1 << 14) +#define RT5639_M_MONO_ADC_L1_SFT 14 +#define RT5639_M_MONO_ADC_L2 (0x1 << 13) +#define RT5639_M_MONO_ADC_L2_SFT 13 +#define RT5639_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5639_MONO_ADC_L1_SRC_SFT 12 +#define RT5639_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5639_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5639_MONO_ADC_L2_SRC_MASK (0x3 << 10) +#define RT5639_MONO_ADC_L2_SRC_SFT 10 +#define RT5639_MONO_ADC_L2_SRC_DMIC_L1 (0x0 << 10) +#define RT5639_MONO_ADC_L2_SRC_DMIC_L2 (0x1 << 10) +#define RT5639_MONO_ADC_L2_SRC_DACMIXL (0x2 << 10) +#define RT5639_M_MONO_ADC_R1 (0x1 << 6) +#define RT5639_M_MONO_ADC_R1_SFT 6 +#define RT5639_M_MONO_ADC_R2 (0x1 << 5) +#define RT5639_M_MONO_ADC_R2_SFT 5 +#define RT5639_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5639_MONO_ADC_R1_SRC_SFT 4 +#define RT5639_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5639_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5639_MONO_ADC_R2_SRC_MASK (0x3 << 2) +#define RT5639_MONO_ADC_R2_SRC_SFT 2 +#define RT5639_MONO_ADC_R2_SRC_DMIC_R1 (0x0 << 2) +#define RT5639_MONO_ADC_R2_SRC_DMIC_R2 (0x1 << 2) +#define RT5639_MONO_ADC_R2_SRC_DACMIXR (0x2 << 2) + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5639_M_ADCMIX_L (0x1 << 15) +#define RT5639_M_ADCMIX_L_SFT 15 +#define RT5639_M_IF1_DAC_L (0x1 << 14) +#define RT5639_M_IF1_DAC_L_SFT 14 +#define RT5639_M_ADCMIX_R (0x1 << 7) +#define RT5639_M_ADCMIX_R_SFT 7 +#define RT5639_M_IF1_DAC_R (0x1 << 6) +#define RT5639_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5639_M_DAC_L1 (0x1 << 14) +#define RT5639_M_DAC_L1_SFT 14 +#define RT5639_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5639_DAC_L1_STO_L_VOL_SFT 13 +#define RT5639_M_DAC_L2 (0x1 << 12) +#define RT5639_M_DAC_L2_SFT 12 +#define RT5639_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5639_DAC_L2_STO_L_VOL_SFT 11 +#define RT5639_M_DAC_R1 (0x1 << 6) +#define RT5639_M_DAC_R1_SFT 6 +#define RT5639_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5639_DAC_R1_STO_R_VOL_SFT 5 +#define RT5639_M_DAC_R2 (0x1 << 4) +#define RT5639_M_DAC_R2_SFT 4 +#define RT5639_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5639_DAC_R2_STO_R_VOL_SFT 3 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5639_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5639_M_DAC_L1_MONO_L_SFT 14 +#define RT5639_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5639_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5639_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5639_M_DAC_L2_MONO_L_SFT 12 +#define RT5639_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5639_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5639_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5639_M_DAC_R2_MONO_L_SFT 10 +#define RT5639_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5639_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5639_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5639_M_DAC_R1_MONO_R_SFT 6 +#define RT5639_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5639_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5639_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5639_M_DAC_R2_MONO_R_SFT 4 +#define RT5639_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5639_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5639_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5639_M_DAC_L2_MONO_R_SFT 2 +#define RT5639_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5639_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5639_M_STO_L_DAC_L (0x1 << 15) +#define RT5639_M_STO_L_DAC_L_SFT 15 +#define RT5639_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5639_STO_L_DAC_L_VOL_SFT 14 +#define RT5639_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5639_M_DAC_L2_DAC_L_SFT 13 +#define RT5639_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5639_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5639_M_STO_R_DAC_R (0x1 << 11) +#define RT5639_M_STO_R_DAC_R_SFT 11 +#define RT5639_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5639_STO_R_DAC_R_VOL_SFT 10 +#define RT5639_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5639_M_DAC_R2_DAC_R_SFT 9 +#define RT5639_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5639_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 2 (0x2e) */ +#define RT5639_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5639_DAC_L2_SEL_SFT 14 +#define RT5639_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5639_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5639_DAC_R2_SEL_SFT 12 +#define RT5639_DAC_R2_SEL_IF2 (0x0 << 12) + +/* Digital Interface Data Control (0x2f) */ +#define RT5639_IF1_DAC_SEL_MASK (0x3 << 14) +#define RT5639_IF1_DAC_SEL_SFT 14 +#define RT5639_IF1_DAC_SEL_NOR (0x0 << 14) +#define RT5639_IF1_DAC_SEL_L2R (0x1 << 14) +#define RT5639_IF1_DAC_SEL_R2L (0x2 << 14) +#define RT5639_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5639_IF1_ADC_SEL_MASK (0x3 << 12) +#define RT5639_IF1_ADC_SEL_SFT 12 +#define RT5639_IF1_ADC_SEL_NOR (0x0 << 12) +#define RT5639_IF1_ADC_SEL_L2R (0x1 << 12) +#define RT5639_IF1_ADC_SEL_R2L (0x2 << 12) +#define RT5639_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5639_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5639_IF2_DAC_SEL_SFT 10 +#define RT5639_IF2_DAC_SEL_NOR (0x0 << 10) +#define RT5639_IF2_DAC_SEL_L2R (0x1 << 10) +#define RT5639_IF2_DAC_SEL_R2L (0x2 << 10) +#define RT5639_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5639_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5639_IF2_ADC_SEL_SFT 8 +#define RT5639_IF2_ADC_SEL_NOR (0x0 << 8) +#define RT5639_IF2_ADC_SEL_L2R (0x1 << 8) +#define RT5639_IF2_ADC_SEL_R2L (0x2 << 8) +#define RT5639_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5639_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5639_IF3_DAC_SEL_SFT 6 +#define RT5639_IF3_DAC_SEL_NOR (0x0 << 6) +#define RT5639_IF3_DAC_SEL_L2R (0x1 << 6) +#define RT5639_IF3_DAC_SEL_R2L (0x2 << 6) +#define RT5639_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5639_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5639_IF3_ADC_SEL_SFT 4 +#define RT5639_IF3_ADC_SEL_NOR (0x0 << 4) +#define RT5639_IF3_ADC_SEL_L2R (0x1 << 4) +#define RT5639_IF3_ADC_SEL_R2L (0x2 << 4) +#define RT5639_IF3_ADC_SEL_SWAP (0x3 << 4) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5639_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5639_G_HP_L_RM_L_SFT 13 +#define RT5639_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5639_G_IN_L_RM_L_SFT 10 +#define RT5639_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5639_G_BST4_RM_L_SFT 7 +#define RT5639_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5639_G_BST3_RM_L_SFT 4 +#define RT5639_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5639_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5639_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5639_G_BST1_RM_L_SFT 13 +#define RT5639_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5639_G_OM_L_RM_L_SFT 10 +#define RT5639_M_HP_L_RM_L (0x1 << 6) +#define RT5639_M_HP_L_RM_L_SFT 6 +#define RT5639_M_IN_L_RM_L (0x1 << 5) +#define RT5639_M_IN_L_RM_L_SFT 5 +#define RT5639_M_BST4_RM_L (0x1 << 4) +#define RT5639_M_BST4_RM_L_SFT 4 +#define RT5639_M_BST3_RM_L (0x1 << 3) +#define RT5639_M_BST3_RM_L_SFT 3 +#define RT5639_M_BST2_RM_L (0x1 << 2) +#define RT5639_M_BST2_RM_L_SFT 2 +#define RT5639_M_BST1_RM_L (0x1 << 1) +#define RT5639_M_BST1_RM_L_SFT 1 +#define RT5639_M_OM_L_RM_L (0x1) +#define RT5639_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5639_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5639_G_HP_R_RM_R_SFT 13 +#define RT5639_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5639_G_IN_R_RM_R_SFT 10 +#define RT5639_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5639_G_BST4_RM_R_SFT 7 +#define RT5639_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5639_G_BST3_RM_R_SFT 4 +#define RT5639_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5639_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5639_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5639_G_BST1_RM_R_SFT 13 +#define RT5639_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5639_G_OM_R_RM_R_SFT 10 +#define RT5639_M_HP_R_RM_R (0x1 << 6) +#define RT5639_M_HP_R_RM_R_SFT 6 +#define RT5639_M_IN_R_RM_R (0x1 << 5) +#define RT5639_M_IN_R_RM_R_SFT 5 +#define RT5639_M_BST4_RM_R (0x1 << 4) +#define RT5639_M_BST4_RM_R_SFT 4 +#define RT5639_M_BST3_RM_R (0x1 << 3) +#define RT5639_M_BST3_RM_R_SFT 3 +#define RT5639_M_BST2_RM_R (0x1 << 2) +#define RT5639_M_BST2_RM_R_SFT 2 +#define RT5639_M_BST1_RM_R (0x1 << 1) +#define RT5639_M_BST1_RM_R_SFT 1 +#define RT5639_M_OM_R_RM_R (0x1) +#define RT5639_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5639_M_DAC2_HM (0x1 << 15) +#define RT5639_M_DAC2_HM_SFT 15 +#define RT5639_M_DAC1_HM (0x1 << 14) +#define RT5639_M_DAC1_HM_SFT 14 +#define RT5639_M_HPVOL_HM (0x1 << 13) +#define RT5639_M_HPVOL_HM_SFT 13 +#define RT5639_G_HPOMIX_MASK (0x1 << 12) +#define RT5639_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5639_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5639_G_RM_L_SM_L_SFT 14 +#define RT5639_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5639_G_IN_L_SM_L_SFT 12 +#define RT5639_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5639_G_DAC_L1_SM_L_SFT 10 +#define RT5639_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5639_G_DAC_L2_SM_L_SFT 8 +#define RT5639_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5639_G_OM_L_SM_L_SFT 6 +#define RT5639_M_RM_L_SM_L (0x1 << 5) +#define RT5639_M_RM_L_SM_L_SFT 5 +#define RT5639_M_IN_L_SM_L (0x1 << 4) +#define RT5639_M_IN_L_SM_L_SFT 4 +#define RT5639_M_DAC_L1_SM_L (0x1 << 3) +#define RT5639_M_DAC_L1_SM_L_SFT 3 +#define RT5639_M_DAC_L2_SM_L (0x1 << 2) +#define RT5639_M_DAC_L2_SM_L_SFT 2 +#define RT5639_M_OM_L_SM_L (0x1 << 1) +#define RT5639_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5639_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5639_G_RM_R_SM_R_SFT 14 +#define RT5639_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5639_G_IN_R_SM_R_SFT 12 +#define RT5639_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5639_G_DAC_R1_SM_R_SFT 10 +#define RT5639_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5639_G_DAC_R2_SM_R_SFT 8 +#define RT5639_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5639_G_OM_R_SM_R_SFT 6 +#define RT5639_M_RM_R_SM_R (0x1 << 5) +#define RT5639_M_RM_R_SM_R_SFT 5 +#define RT5639_M_IN_R_SM_R (0x1 << 4) +#define RT5639_M_IN_R_SM_R_SFT 4 +#define RT5639_M_DAC_R1_SM_R (0x1 << 3) +#define RT5639_M_DAC_R1_SM_R_SFT 3 +#define RT5639_M_DAC_R2_SM_R (0x1 << 2) +#define RT5639_M_DAC_R2_SM_R_SFT 2 +#define RT5639_M_OM_R_SM_R (0x1 << 1) +#define RT5639_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5639_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5639_M_DAC_R1_SPM_L_SFT 15 +#define RT5639_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5639_M_DAC_L1_SPM_L_SFT 14 +#define RT5639_M_SV_R_SPM_L (0x1 << 13) +#define RT5639_M_SV_R_SPM_L_SFT 13 +#define RT5639_M_SV_L_SPM_L (0x1 << 12) +#define RT5639_M_SV_L_SPM_L_SFT 12 +#define RT5639_M_BST1_SPM_L (0x1 << 11) +#define RT5639_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5639_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5639_M_DAC_R1_SPM_R_SFT 13 +#define RT5639_M_SV_R_SPM_R (0x1 << 12) +#define RT5639_M_SV_R_SPM_R_SFT 12 +#define RT5639_M_BST1_SPM_R (0x1 << 11) +#define RT5639_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5639_SPO_CLSD_RATIO_MASK (0x7) +#define RT5639_SPO_CLSD_RATIO_SFT 0 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5639_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5639_G_BST3_OM_L_SFT 13 +#define RT5639_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5639_G_BST2_OM_L_SFT 10 +#define RT5639_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5639_G_BST1_OM_L_SFT 7 +#define RT5639_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5639_G_IN_L_OM_L_SFT 4 +#define RT5639_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5639_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5639_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5639_G_DAC_R2_OM_L_SFT 13 +#define RT5639_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5639_G_DAC_L2_OM_L_SFT 10 +#define RT5639_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5639_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5639_M_BST3_OM_L (0x1 << 7) +#define RT5639_M_BST3_OM_L_SFT 7 +#define RT5639_M_BST2_OM_L (0x1 << 6) +#define RT5639_M_BST2_OM_L_SFT 6 +#define RT5639_M_BST1_OM_L (0x1 << 5) +#define RT5639_M_BST1_OM_L_SFT 5 +#define RT5639_M_IN_L_OM_L (0x1 << 4) +#define RT5639_M_IN_L_OM_L_SFT 4 +#define RT5639_M_RM_L_OM_L (0x1 << 3) +#define RT5639_M_RM_L_OM_L_SFT 3 +#define RT5639_M_DAC_R2_OM_L (0x1 << 2) +#define RT5639_M_DAC_R2_OM_L_SFT 2 +#define RT5639_M_DAC_L2_OM_L (0x1 << 1) +#define RT5639_M_DAC_L2_OM_L_SFT 1 +#define RT5639_M_DAC_L1_OM_L (0x1) +#define RT5639_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5639_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5639_G_BST4_OM_R_SFT 13 +#define RT5639_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5639_G_BST2_OM_R_SFT 10 +#define RT5639_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5639_G_BST1_OM_R_SFT 7 +#define RT5639_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5639_G_IN_R_OM_R_SFT 4 +#define RT5639_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5639_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5639_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5639_G_DAC_L2_OM_R_SFT 13 +#define RT5639_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5639_G_DAC_R2_OM_R_SFT 10 +#define RT5639_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5639_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5639_M_BST4_OM_R (0x1 << 7) +#define RT5639_M_BST4_OM_R_SFT 7 +#define RT5639_M_BST2_OM_R (0x1 << 6) +#define RT5639_M_BST2_OM_R_SFT 6 +#define RT5639_M_BST1_OM_R (0x1 << 5) +#define RT5639_M_BST1_OM_R_SFT 5 +#define RT5639_M_IN_R_OM_R (0x1 << 4) +#define RT5639_M_IN_R_OM_R_SFT 4 +#define RT5639_M_RM_R_OM_R (0x1 << 3) +#define RT5639_M_RM_R_OM_R_SFT 3 +#define RT5639_M_DAC_L2_OM_R (0x1 << 2) +#define RT5639_M_DAC_L2_OM_R_SFT 2 +#define RT5639_M_DAC_R2_OM_R (0x1 << 1) +#define RT5639_M_DAC_R2_OM_R_SFT 1 +#define RT5639_M_DAC_R1_OM_R (0x1) +#define RT5639_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5639_M_DAC_L1_LM (0x1 << 15) +#define RT5639_M_DAC_L1_LM_SFT 15 +#define RT5639_M_DAC_R1_LM (0x1 << 14) +#define RT5639_M_DAC_R1_LM_SFT 14 +#define RT5639_M_OV_L_LM (0x1 << 13) +#define RT5639_M_OV_L_LM_SFT 13 +#define RT5639_M_OV_R_LM (0x1 << 12) +#define RT5639_M_OV_R_LM_SFT 12 +#define RT5639_G_LOUTMIX_MASK (0x1 << 11) +#define RT5639_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5639_PWR_I2S1 (0x1 << 15) +#define RT5639_PWR_I2S1_BIT 15 +#define RT5639_PWR_I2S2 (0x1 << 14) +#define RT5639_PWR_I2S2_BIT 14 +#define RT5639_PWR_DAC_L1 (0x1 << 12) +#define RT5639_PWR_DAC_L1_BIT 12 +#define RT5639_PWR_DAC_R1 (0x1 << 11) +#define RT5639_PWR_DAC_R1_BIT 11 +#define RT5639_PWR_DAC_L2 (0x1 << 7) +#define RT5639_PWR_DAC_L2_BIT 7 +#define RT5639_PWR_DAC_R2 (0x1 << 6) +#define RT5639_PWR_DAC_R2_BIT 6 +#define RT5639_PWR_ADC_L (0x1 << 2) +#define RT5639_PWR_ADC_L_BIT 2 +#define RT5639_PWR_ADC_R (0x1 << 1) +#define RT5639_PWR_ADC_R_BIT 1 +#define RT5639_PWR_CLS_D (0x1) +#define RT5639_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5639_PWR_ADC_SF (0x1 << 15) +#define RT5639_PWR_ADC_SF_BIT 15 +#define RT5639_PWR_ADC_MF_L (0x1 << 14) +#define RT5639_PWR_ADC_MF_L_BIT 14 +#define RT5639_PWR_ADC_MF_R (0x1 << 13) +#define RT5639_PWR_ADC_MF_R_BIT 13 +#define RT5639_PWR_I2S_DSP (0x1 << 12) +#define RT5639_PWR_I2S_DSP_BIT 12 + +/* Power Management for Analog 1 (0x63) */ +#define RT5639_PWR_VREF1 (0x1 << 15) +#define RT5639_PWR_VREF1_BIT 15 +#define RT5639_PWR_FV1 (0x1 << 14) +#define RT5639_PWR_FV1_BIT 14 +#define RT5639_PWR_MB (0x1 << 13) +#define RT5639_PWR_MB_BIT 13 +#define RT5639_PWR_LM (0x1 << 12) +#define RT5639_PWR_LM_BIT 12 +#define RT5639_PWR_BG (0x1 << 11) +#define RT5639_PWR_BG_BIT 11 +#define RT5639_PWR_MM (0x1 << 10) +#define RT5639_PWR_MM_BIT 10 +#define RT5639_PWR_MA (0x1 << 8) +#define RT5639_PWR_MA_BIT 8 +#define RT5639_PWR_HP_L (0x1 << 7) +#define RT5639_PWR_HP_L_BIT 7 +#define RT5639_PWR_HP_R (0x1 << 6) +#define RT5639_PWR_HP_R_BIT 6 +#define RT5639_PWR_HA (0x1 << 5) +#define RT5639_PWR_HA_BIT 5 +#define RT5639_PWR_VREF2 (0x1 << 4) +#define RT5639_PWR_VREF2_BIT 4 +#define RT5639_PWR_FV2 (0x1 << 3) +#define RT5639_PWR_FV2_BIT 3 +#define RT5639_PWR_LDO2 (0x1 << 2) +#define RT5639_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5639_PWR_BST1 (0x1 << 15) +#define RT5639_PWR_BST1_BIT 15 +#define RT5639_PWR_BST2 (0x1 << 14) +#define RT5639_PWR_BST2_BIT 14 +#define RT5639_PWR_BST3 (0x1 << 13) +#define RT5639_PWR_BST3_BIT 13 +#define RT5639_PWR_BST4 (0x1 << 12) +#define RT5639_PWR_BST4_BIT 12 +#define RT5639_PWR_MB1 (0x1 << 11) +#define RT5639_PWR_MB1_BIT 11 +#define RT5639_PWR_PLL (0x1 << 9) +#define RT5639_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5639_PWR_OM_L (0x1 << 15) +#define RT5639_PWR_OM_L_BIT 15 +#define RT5639_PWR_OM_R (0x1 << 14) +#define RT5639_PWR_OM_R_BIT 14 +#define RT5639_PWR_SM_L (0x1 << 13) +#define RT5639_PWR_SM_L_BIT 13 +#define RT5639_PWR_SM_R (0x1 << 12) +#define RT5639_PWR_SM_R_BIT 12 +#define RT5639_PWR_RM_L (0x1 << 11) +#define RT5639_PWR_RM_L_BIT 11 +#define RT5639_PWR_RM_R (0x1 << 10) +#define RT5639_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5639_PWR_SV_L (0x1 << 15) +#define RT5639_PWR_SV_L_BIT 15 +#define RT5639_PWR_SV_R (0x1 << 14) +#define RT5639_PWR_SV_R_BIT 14 +#define RT5639_PWR_OV_L (0x1 << 13) +#define RT5639_PWR_OV_L_BIT 13 +#define RT5639_PWR_OV_R (0x1 << 12) +#define RT5639_PWR_OV_R_BIT 12 +#define RT5639_PWR_HV_L (0x1 << 11) +#define RT5639_PWR_HV_L_BIT 11 +#define RT5639_PWR_HV_R (0x1 << 10) +#define RT5639_PWR_HV_R_BIT 10 +#define RT5639_PWR_IN_L (0x1 << 9) +#define RT5639_PWR_IN_L_BIT 9 +#define RT5639_PWR_IN_R (0x1 << 8) +#define RT5639_PWR_IN_R_BIT 8 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5639_I2S_MS_MASK (0x1 << 15) +#define RT5639_I2S_MS_SFT 15 +#define RT5639_I2S_MS_M (0x0 << 15) +#define RT5639_I2S_MS_S (0x1 << 15) +#define RT5639_I2S_IF_MASK (0x7 << 12) +#define RT5639_I2S_IF_SFT 12 +#define RT5639_I2S_O_CP_MASK (0x3 << 10) +#define RT5639_I2S_O_CP_SFT 10 +#define RT5639_I2S_O_CP_OFF (0x0 << 10) +#define RT5639_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5639_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5639_I2S_I_CP_MASK (0x3 << 8) +#define RT5639_I2S_I_CP_SFT 8 +#define RT5639_I2S_I_CP_OFF (0x0 << 8) +#define RT5639_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5639_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5639_I2S_BP_MASK (0x1 << 7) +#define RT5639_I2S_BP_SFT 7 +#define RT5639_I2S_BP_NOR (0x0 << 7) +#define RT5639_I2S_BP_INV (0x1 << 7) +#define RT5639_I2S_DL_MASK (0x3 << 2) +#define RT5639_I2S_DL_SFT 2 +#define RT5639_I2S_DL_16 (0x0 << 2) +#define RT5639_I2S_DL_20 (0x1 << 2) +#define RT5639_I2S_DL_24 (0x2 << 2) +#define RT5639_I2S_DL_8 (0x3 << 2) +#define RT5639_I2S_DF_MASK (0x3) +#define RT5639_I2S_DF_SFT 0 +#define RT5639_I2S_DF_I2S (0x0) +#define RT5639_I2S_DF_LEFT (0x1) +#define RT5639_I2S_DF_PCM_A (0x2) +#define RT5639_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5639_I2S2_SDI_MASK (0x1 << 6) +#define RT5639_I2S2_SDI_SFT 6 +#define RT5639_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5639_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5639_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5639_I2S_BCLK_MS1_SFT 15 +#define RT5639_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5639_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5639_I2S_PD1_MASK (0x7 << 12) +#define RT5639_I2S_PD1_SFT 12 +#define RT5639_I2S_PD1_1 (0x0 << 12) +#define RT5639_I2S_PD1_2 (0x1 << 12) +#define RT5639_I2S_PD1_3 (0x2 << 12) +#define RT5639_I2S_PD1_4 (0x3 << 12) +#define RT5639_I2S_PD1_6 (0x4 << 12) +#define RT5639_I2S_PD1_8 (0x5 << 12) +#define RT5639_I2S_PD1_12 (0x6 << 12) +#define RT5639_I2S_PD1_16 (0x7 << 12) +#define RT5639_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5639_I2S_BCLK_MS2_SFT 11 +#define RT5639_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5639_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5639_I2S_PD2_MASK (0x7 << 8) +#define RT5639_I2S_PD2_SFT 8 +#define RT5639_I2S_PD2_1 (0x0 << 8) +#define RT5639_I2S_PD2_2 (0x1 << 8) +#define RT5639_I2S_PD2_3 (0x2 << 8) +#define RT5639_I2S_PD2_4 (0x3 << 8) +#define RT5639_I2S_PD2_6 (0x4 << 8) +#define RT5639_I2S_PD2_8 (0x5 << 8) +#define RT5639_I2S_PD2_12 (0x6 << 8) +#define RT5639_I2S_PD2_16 (0x7 << 8) +#define RT5639_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5639_I2S_BCLK_MS3_SFT 7 +#define RT5639_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5639_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5639_I2S_PD3_MASK (0x7 << 4) +#define RT5639_I2S_PD3_SFT 4 +#define RT5639_I2S_PD3_1 (0x0 << 4) +#define RT5639_I2S_PD3_2 (0x1 << 4) +#define RT5639_I2S_PD3_3 (0x2 << 4) +#define RT5639_I2S_PD3_4 (0x3 << 4) +#define RT5639_I2S_PD3_6 (0x4 << 4) +#define RT5639_I2S_PD3_8 (0x5 << 4) +#define RT5639_I2S_PD3_12 (0x6 << 4) +#define RT5639_I2S_PD3_16 (0x7 << 4) +#define RT5639_DAC_OSR_MASK (0x3 << 2) +#define RT5639_DAC_OSR_SFT 2 +#define RT5639_DAC_OSR_128 (0x0 << 2) +#define RT5639_DAC_OSR_64 (0x1 << 2) +#define RT5639_DAC_OSR_32 (0x2 << 2) +#define RT5639_DAC_OSR_16 (0x3 << 2) +#define RT5639_ADC_OSR_MASK (0x3) +#define RT5639_ADC_OSR_SFT 0 +#define RT5639_ADC_OSR_128 (0x0) +#define RT5639_ADC_OSR_64 (0x1) +#define RT5639_ADC_OSR_32 (0x2) +#define RT5639_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5639_DAC_L_OSR_MASK (0x3 << 14) +#define RT5639_DAC_L_OSR_SFT 14 +#define RT5639_DAC_L_OSR_128 (0x0 << 14) +#define RT5639_DAC_L_OSR_64 (0x1 << 14) +#define RT5639_DAC_L_OSR_32 (0x2 << 14) +#define RT5639_DAC_L_OSR_16 (0x3 << 14) +#define RT5639_ADC_R_OSR_MASK (0x3 << 12) +#define RT5639_ADC_R_OSR_SFT 12 +#define RT5639_ADC_R_OSR_128 (0x0 << 12) +#define RT5639_ADC_R_OSR_64 (0x1 << 12) +#define RT5639_ADC_R_OSR_32 (0x2 << 12) +#define RT5639_ADC_R_OSR_16 (0x3 << 12) +#define RT5639_DAHPF_EN (0x1 << 11) +#define RT5639_DAHPF_EN_SFT 11 +#define RT5639_ADHPF_EN (0x1 << 10) +#define RT5639_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5639_DMIC_1_EN_MASK (0x1 << 15) +#define RT5639_DMIC_1_EN_SFT 15 +#define RT5639_DMIC_1_DIS (0x0 << 15) +#define RT5639_DMIC_1_EN (0x1 << 15) +#define RT5639_DMIC_2_EN_MASK (0x1 << 14) +#define RT5639_DMIC_2_EN_SFT 14 +#define RT5639_DMIC_2_DIS (0x0 << 14) +#define RT5639_DMIC_2_EN (0x1 << 14) +#define RT5639_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5639_DMIC_1L_LH_SFT 13 +#define RT5639_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5639_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5639_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5639_DMIC_1R_LH_SFT 12 +#define RT5639_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5639_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5639_DMIC_1_DP_MASK (0x1 << 11) +#define RT5639_DMIC_1_DP_SFT 11 +#define RT5639_DMIC_1_DP_GPIO3 (0x0 << 11) +#define RT5639_DMIC_1_DP_IN1P (0x1 << 11) +#define RT5639_DMIC_2_DP_MASK (0x1 << 10) +#define RT5639_DMIC_2_DP_SFT 10 +#define RT5639_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5639_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5639_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5639_DMIC_2L_LH_SFT 9 +#define RT5639_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5639_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5639_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5639_DMIC_2R_LH_SFT 8 +#define RT5639_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5639_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5639_DMIC_CLK_MASK (0x7 << 5) +#define RT5639_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5639_SCLK_SRC_MASK (0x3 << 14) +#define RT5639_SCLK_SRC_SFT 14 +#define RT5639_SCLK_SRC_MCLK (0x0 << 14) +#define RT5639_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5639_PLL1_SRC_MASK (0x3 << 12) +#define RT5639_PLL1_SRC_SFT 12 +#define RT5639_PLL1_SRC_MCLK (0x0 << 12) +#define RT5639_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5639_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5639_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5639_PLL1_PD_MASK (0x1 << 3) +#define RT5639_PLL1_PD_SFT 3 +#define RT5639_PLL1_PD_1 (0x0 << 3) +#define RT5639_PLL1_PD_2 (0x1 << 3) + +#define RT5639_PLL_INP_MAX 40000000 +#define RT5639_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5639_PLL_N_MAX 0x1ff +#define RT5639_PLL_N_MASK (RT5639_PLL_N_MAX << 7) +#define RT5639_PLL_N_SFT 7 +#define RT5639_PLL_K_MAX 0x1f +#define RT5639_PLL_K_MASK (RT5639_PLL_K_MAX) +#define RT5639_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5639_PLL_M_MAX 0xf +#define RT5639_PLL_M_MASK (RT5639_PLL_M_MAX << 12) +#define RT5639_PLL_M_SFT 12 +#define RT5639_PLL_M_BP (0x1 << 11) +#define RT5639_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5639_STO_T_MASK (0x1 << 15) +#define RT5639_STO_T_SFT 15 +#define RT5639_STO_T_SCLK (0x0 << 15) +#define RT5639_STO_T_LRCK1 (0x1 << 15) +#define RT5639_M1_T_MASK (0x1 << 14) +#define RT5639_M1_T_SFT 14 +#define RT5639_M1_T_I2S2 (0x0 << 14) +#define RT5639_M1_T_I2S2_D3 (0x1 << 14) +#define RT5639_I2S2_F_MASK (0x1 << 12) +#define RT5639_I2S2_F_SFT 12 +#define RT5639_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5639_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5639_DMIC_1_M_MASK (0x1 << 9) +#define RT5639_DMIC_1_M_SFT 9 +#define RT5639_DMIC_1_M_NOR (0x0 << 9) +#define RT5639_DMIC_1_M_ASYN (0x1 << 9) +#define RT5639_DMIC_2_M_MASK (0x1 << 8) +#define RT5639_DMIC_2_M_SFT 8 +#define RT5639_DMIC_2_M_NOR (0x0 << 8) +#define RT5639_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5639_MDA_L_M_MASK (0x1 << 15) +#define RT5639_MDA_L_M_SFT 15 +#define RT5639_MDA_L_M_NOR (0x0 << 15) +#define RT5639_MDA_L_M_ASYN (0x1 << 15) +#define RT5639_MDA_R_M_MASK (0x1 << 14) +#define RT5639_MDA_R_M_SFT 14 +#define RT5639_MDA_R_M_NOR (0x0 << 14) +#define RT5639_MDA_R_M_ASYN (0x1 << 14) +#define RT5639_MAD_L_M_MASK (0x1 << 13) +#define RT5639_MAD_L_M_SFT 13 +#define RT5639_MAD_L_M_NOR (0x0 << 13) +#define RT5639_MAD_L_M_ASYN (0x1 << 13) +#define RT5639_MAD_R_M_MASK (0x1 << 12) +#define RT5639_MAD_R_M_SFT 12 +#define RT5639_MAD_R_M_NOR (0x0 << 12) +#define RT5639_MAD_R_M_ASYN (0x1 << 12) +#define RT5639_ADC_M_MASK (0x1 << 11) +#define RT5639_ADC_M_SFT 11 +#define RT5639_ADC_M_NOR (0x0 << 11) +#define RT5639_ADC_M_ASYN (0x1 << 11) +#define RT5639_STO_DAC_M_MASK (0x1 << 5) +#define RT5639_STO_DAC_M_SFT 5 +#define RT5639_STO_DAC_M_NOR (0x0 << 5) +#define RT5639_STO_DAC_M_ASYN (0x1 << 5) +#define RT5639_I2S1_R_D_MASK (0x1 << 4) +#define RT5639_I2S1_R_D_SFT 4 +#define RT5639_I2S1_R_D_DIS (0x0 << 4) +#define RT5639_I2S1_R_D_EN (0x1 << 4) +#define RT5639_I2S2_R_D_MASK (0x1 << 3) +#define RT5639_I2S2_R_D_SFT 3 +#define RT5639_I2S2_R_D_DIS (0x0 << 3) +#define RT5639_I2S2_R_D_EN (0x1 << 3) +#define RT5639_PRE_SCLK_MASK (0x3) +#define RT5639_PRE_SCLK_SFT 0 +#define RT5639_PRE_SCLK_512 (0x0) +#define RT5639_PRE_SCLK_1024 (0x1) +#define RT5639_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5639_I2S1_RATE_MASK (0xf << 12) +#define RT5639_I2S1_RATE_SFT 12 +#define RT5639_I2S2_RATE_MASK (0xf << 8) +#define RT5639_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5639_I2S1_PD_MASK (0x7 << 12) +#define RT5639_I2S1_PD_SFT 12 +#define RT5639_I2S2_PD_MASK (0x7 << 8) +#define RT5639_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5639_HP_OVCD_MASK (0x1 << 10) +#define RT5639_HP_OVCD_SFT 10 +#define RT5639_HP_OVCD_DIS (0x0 << 10) +#define RT5639_HP_OVCD_EN (0x1 << 10) +#define RT5639_HP_OC_TH_MASK (0x3 << 8) +#define RT5639_HP_OC_TH_SFT 8 +#define RT5639_HP_OC_TH_90 (0x0 << 8) +#define RT5639_HP_OC_TH_105 (0x1 << 8) +#define RT5639_HP_OC_TH_120 (0x2 << 8) +#define RT5639_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5639_CLSD_OC_MASK (0x1 << 9) +#define RT5639_CLSD_OC_SFT 9 +#define RT5639_CLSD_OC_PU (0x0 << 9) +#define RT5639_CLSD_OC_PD (0x1 << 9) +#define RT5639_AUTO_PD_MASK (0x1 << 8) +#define RT5639_AUTO_PD_SFT 8 +#define RT5639_AUTO_PD_DIS (0x0 << 8) +#define RT5639_AUTO_PD_EN (0x1 << 8) +#define RT5639_CLSD_OC_TH_MASK (0x3f) +#define RT5639_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5639_CLSD_RATIO_MASK (0xf << 12) +#define RT5639_CLSD_RATIO_SFT 12 +#define RT5639_CLSD_OM_MASK (0x1 << 11) +#define RT5639_CLSD_OM_SFT 11 +#define RT5639_CLSD_OM_MONO (0x0 << 11) +#define RT5639_CLSD_OM_STO (0x1 << 11) +#define RT5639_CLSD_SCH_MASK (0x1 << 10) +#define RT5639_CLSD_SCH_SFT 10 +#define RT5639_CLSD_SCH_L (0x0 << 10) +#define RT5639_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5639_SMT_TRIG_MASK (0x1 << 15) +#define RT5639_SMT_TRIG_SFT 15 +#define RT5639_SMT_TRIG_DIS (0x0 << 15) +#define RT5639_SMT_TRIG_EN (0x1 << 15) +#define RT5639_HP_L_SMT_MASK (0x1 << 9) +#define RT5639_HP_L_SMT_SFT 9 +#define RT5639_HP_L_SMT_DIS (0x0 << 9) +#define RT5639_HP_L_SMT_EN (0x1 << 9) +#define RT5639_HP_R_SMT_MASK (0x1 << 8) +#define RT5639_HP_R_SMT_SFT 8 +#define RT5639_HP_R_SMT_DIS (0x0 << 8) +#define RT5639_HP_R_SMT_EN (0x1 << 8) +#define RT5639_HP_CD_PD_MASK (0x1 << 7) +#define RT5639_HP_CD_PD_SFT 7 +#define RT5639_HP_CD_PD_DIS (0x0 << 7) +#define RT5639_HP_CD_PD_EN (0x1 << 7) +#define RT5639_RSTN_MASK (0x1 << 6) +#define RT5639_RSTN_SFT 6 +#define RT5639_RSTN_DIS (0x0 << 6) +#define RT5639_RSTN_EN (0x1 << 6) +#define RT5639_RSTP_MASK (0x1 << 5) +#define RT5639_RSTP_SFT 5 +#define RT5639_RSTP_DIS (0x0 << 5) +#define RT5639_RSTP_EN (0x1 << 5) +#define RT5639_HP_CO_MASK (0x1 << 4) +#define RT5639_HP_CO_SFT 4 +#define RT5639_HP_CO_DIS (0x0 << 4) +#define RT5639_HP_CO_EN (0x1 << 4) +#define RT5639_HP_CP_MASK (0x1 << 3) +#define RT5639_HP_CP_SFT 3 +#define RT5639_HP_CP_PD (0x0 << 3) +#define RT5639_HP_CP_PU (0x1 << 3) +#define RT5639_HP_SG_MASK (0x1 << 2) +#define RT5639_HP_SG_SFT 2 +#define RT5639_HP_SG_DIS (0x0 << 2) +#define RT5639_HP_SG_EN (0x1 << 2) +#define RT5639_HP_DP_MASK (0x1 << 1) +#define RT5639_HP_DP_SFT 1 +#define RT5639_HP_DP_PD (0x0 << 1) +#define RT5639_HP_DP_PU (0x1 << 1) +#define RT5639_HP_CB_MASK (0x1) +#define RT5639_HP_CB_SFT 0 +#define RT5639_HP_CB_PD (0x0) +#define RT5639_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5639_DEPOP_MASK (0x1 << 13) +#define RT5639_DEPOP_SFT 13 +#define RT5639_DEPOP_AUTO (0x0 << 13) +#define RT5639_DEPOP_MAN (0x1 << 13) +#define RT5639_RAMP_MASK (0x1 << 12) +#define RT5639_RAMP_SFT 12 +#define RT5639_RAMP_DIS (0x0 << 12) +#define RT5639_RAMP_EN (0x1 << 12) +#define RT5639_BPS_MASK (0x1 << 11) +#define RT5639_BPS_SFT 11 +#define RT5639_BPS_DIS (0x0 << 11) +#define RT5639_BPS_EN (0x1 << 11) +#define RT5639_FAST_UPDN_MASK (0x1 << 10) +#define RT5639_FAST_UPDN_SFT 10 +#define RT5639_FAST_UPDN_DIS (0x0 << 10) +#define RT5639_FAST_UPDN_EN (0x1 << 10) +#define RT5639_MRES_MASK (0x3 << 8) +#define RT5639_MRES_SFT 8 +#define RT5639_MRES_15MO (0x0 << 8) +#define RT5639_MRES_25MO (0x1 << 8) +#define RT5639_MRES_35MO (0x2 << 8) +#define RT5639_MRES_45MO (0x3 << 8) +#define RT5639_VLO_MASK (0x1 << 7) +#define RT5639_VLO_SFT 7 +#define RT5639_VLO_3V (0x0 << 7) +#define RT5639_VLO_32V (0x1 << 7) +#define RT5639_DIG_DP_MASK (0x1 << 6) +#define RT5639_DIG_DP_SFT 6 +#define RT5639_DIG_DP_DIS (0x0 << 6) +#define RT5639_DIG_DP_EN (0x1 << 6) +#define RT5639_DP_TH_MASK (0x3 << 4) +#define RT5639_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5639_CP_SYS_MASK (0x7 << 12) +#define RT5639_CP_SYS_SFT 12 +#define RT5639_CP_FQ1_MASK (0x7 << 8) +#define RT5639_CP_FQ1_SFT 8 +#define RT5639_CP_FQ2_MASK (0x7 << 4) +#define RT5639_CP_FQ2_SFT 4 +#define RT5639_CP_FQ3_MASK (0x7) +#define RT5639_CP_FQ3_SFT 0 +#define RT5639_CP_FQ_1_5_KHZ 0 +#define RT5639_CP_FQ_3_KHZ 1 +#define RT5639_CP_FQ_6_KHZ 2 +#define RT5639_CP_FQ_12_KHZ 3 +#define RT5639_CP_FQ_24_KHZ 4 +#define RT5639_CP_FQ_48_KHZ 5 +#define RT5639_CP_FQ_96_KHZ 6 +#define RT5639_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5639_OSW_L_MASK (0x1 << 11) +#define RT5639_OSW_L_SFT 11 +#define RT5639_OSW_L_DIS (0x0 << 11) +#define RT5639_OSW_L_EN (0x1 << 11) +#define RT5639_OSW_R_MASK (0x1 << 10) +#define RT5639_OSW_R_SFT 10 +#define RT5639_OSW_R_DIS (0x0 << 10) +#define RT5639_OSW_R_EN (0x1 << 10) +#define RT5639_PM_HP_MASK (0x3 << 8) +#define RT5639_PM_HP_SFT 8 +#define RT5639_PM_HP_LV (0x0 << 8) +#define RT5639_PM_HP_MV (0x1 << 8) +#define RT5639_PM_HP_HV (0x2 << 8) +#define RT5639_IB_HP_MASK (0x3 << 6) +#define RT5639_IB_HP_SFT 6 +#define RT5639_IB_HP_125IL (0x0 << 6) +#define RT5639_IB_HP_25IL (0x1 << 6) +#define RT5639_IB_HP_5IL (0x2 << 6) +#define RT5639_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5639_PVDD_DET_MASK (0x1 << 15) +#define RT5639_PVDD_DET_SFT 15 +#define RT5639_PVDD_DET_DIS (0x0 << 15) +#define RT5639_PVDD_DET_EN (0x1 << 15) +#define RT5639_SPK_AG_MASK (0x1 << 14) +#define RT5639_SPK_AG_SFT 14 +#define RT5639_SPK_AG_DIS (0x0 << 14) +#define RT5639_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5639_MIC1_BS_MASK (0x1 << 15) +#define RT5639_MIC1_BS_SFT 15 +#define RT5639_MIC1_BS_9AV (0x0 << 15) +#define RT5639_MIC1_BS_75AV (0x1 << 15) +#define RT5639_MIC2_BS_MASK (0x1 << 14) +#define RT5639_MIC2_BS_SFT 14 +#define RT5639_MIC2_BS_9AV (0x0 << 14) +#define RT5639_MIC2_BS_75AV (0x1 << 14) +#define RT5639_MIC1_CLK_MASK (0x1 << 13) +#define RT5639_MIC1_CLK_SFT 13 +#define RT5639_MIC1_CLK_DIS (0x0 << 13) +#define RT5639_MIC1_CLK_EN (0x1 << 13) +#define RT5639_MIC2_CLK_MASK (0x1 << 12) +#define RT5639_MIC2_CLK_SFT 12 +#define RT5639_MIC2_CLK_DIS (0x0 << 12) +#define RT5639_MIC2_CLK_EN (0x1 << 12) +#define RT5639_MIC1_OVCD_MASK (0x1 << 11) +#define RT5639_MIC1_OVCD_SFT 11 +#define RT5639_MIC1_OVCD_DIS (0x0 << 11) +#define RT5639_MIC1_OVCD_EN (0x1 << 11) +#define RT5639_MIC1_OVTH_MASK (0x3 << 9) +#define RT5639_MIC1_OVTH_SFT 9 +#define RT5639_MIC1_OVTH_600UA (0x0 << 9) +#define RT5639_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5639_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5639_MIC2_OVCD_MASK (0x1 << 8) +#define RT5639_MIC2_OVCD_SFT 8 +#define RT5639_MIC2_OVCD_DIS (0x0 << 8) +#define RT5639_MIC2_OVCD_EN (0x1 << 8) +#define RT5639_MIC2_OVTH_MASK (0x3 << 6) +#define RT5639_MIC2_OVTH_SFT 6 +#define RT5639_MIC2_OVTH_600UA (0x0 << 6) +#define RT5639_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5639_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5639_PWR_MB_MASK (0x1 << 5) +#define RT5639_PWR_MB_SFT 5 +#define RT5639_PWR_MB_PD (0x0 << 5) +#define RT5639_PWR_MB_PU (0x1 << 5) +#define RT5639_PWR_CLK25M_MASK (0x1 << 4) +#define RT5639_PWR_CLK25M_SFT 4 +#define RT5639_PWR_CLK25M_PD (0x0 << 4) +#define RT5639_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5639_EQ_SRC_MASK (0x1 << 15) +#define RT5639_EQ_SRC_SFT 15 +#define RT5639_EQ_SRC_DAC (0x0 << 15) +#define RT5639_EQ_SRC_ADC (0x1 << 15) +#define RT5639_EQ_UPD (0x1 << 14) +#define RT5639_EQ_UPD_BIT 14 +#define RT5639_EQ_CD_MASK (0x1 << 13) +#define RT5639_EQ_CD_SFT 13 +#define RT5639_EQ_CD_DIS (0x0 << 13) +#define RT5639_EQ_CD_EN (0x1 << 13) +#define RT5639_EQ_DITH_MASK (0x3 << 8) +#define RT5639_EQ_DITH_SFT 8 +#define RT5639_EQ_DITH_NOR (0x0 << 8) +#define RT5639_EQ_DITH_LSB (0x1 << 8) +#define RT5639_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5639_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5639_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5639_EQ_HPF1_M_SFT 8 +#define RT5639_EQ_HPF1_M_HI (0x0 << 8) +#define RT5639_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5639_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5639_EQ_LPF1_M_SFT 7 +#define RT5639_EQ_LPF1_M_LO (0x0 << 7) +#define RT5639_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5639_EQ_HPF2_MASK (0x1 << 6) +#define RT5639_EQ_HPF2_SFT 6 +#define RT5639_EQ_HPF2_DIS (0x0 << 6) +#define RT5639_EQ_HPF2_EN (0x1 << 6) +#define RT5639_EQ_HPF1_MASK (0x1 << 5) +#define RT5639_EQ_HPF1_SFT 5 +#define RT5639_EQ_HPF1_DIS (0x0 << 5) +#define RT5639_EQ_HPF1_EN (0x1 << 5) +#define RT5639_EQ_BPF4_MASK (0x1 << 4) +#define RT5639_EQ_BPF4_SFT 4 +#define RT5639_EQ_BPF4_DIS (0x0 << 4) +#define RT5639_EQ_BPF4_EN (0x1 << 4) +#define RT5639_EQ_BPF3_MASK (0x1 << 3) +#define RT5639_EQ_BPF3_SFT 3 +#define RT5639_EQ_BPF3_DIS (0x0 << 3) +#define RT5639_EQ_BPF3_EN (0x1 << 3) +#define RT5639_EQ_BPF2_MASK (0x1 << 2) +#define RT5639_EQ_BPF2_SFT 2 +#define RT5639_EQ_BPF2_DIS (0x0 << 2) +#define RT5639_EQ_BPF2_EN (0x1 << 2) +#define RT5639_EQ_BPF1_MASK (0x1 << 1) +#define RT5639_EQ_BPF1_SFT 1 +#define RT5639_EQ_BPF1_DIS (0x0 << 1) +#define RT5639_EQ_BPF1_EN (0x1 << 1) +#define RT5639_EQ_LPF_MASK (0x1) +#define RT5639_EQ_LPF_SFT 0 +#define RT5639_EQ_LPF_DIS (0x0) +#define RT5639_EQ_LPF_EN (0x1) + +/* Memory Test (0xb2) */ +#define RT5639_MT_MASK (0x1 << 15) +#define RT5639_MT_SFT 15 +#define RT5639_MT_DIS (0x0 << 15) +#define RT5639_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5639_DRC_AGC_P_MASK (0x1 << 15) +#define RT5639_DRC_AGC_P_SFT 15 +#define RT5639_DRC_AGC_P_DAC (0x0 << 15) +#define RT5639_DRC_AGC_P_ADC (0x1 << 15) +#define RT5639_DRC_AGC_MASK (0x1 << 14) +#define RT5639_DRC_AGC_SFT 14 +#define RT5639_DRC_AGC_DIS (0x0 << 14) +#define RT5639_DRC_AGC_EN (0x1 << 14) +#define RT5639_DRC_AGC_UPD (0x1 << 13) +#define RT5639_DRC_AGC_UPD_BIT 13 +#define RT5639_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5639_DRC_AGC_AR_SFT 8 +#define RT5639_DRC_AGC_R_MASK (0x7 << 5) +#define RT5639_DRC_AGC_R_SFT 5 +#define RT5639_DRC_AGC_R_48K (0x1 << 5) +#define RT5639_DRC_AGC_R_96K (0x2 << 5) +#define RT5639_DRC_AGC_R_192K (0x3 << 5) +#define RT5639_DRC_AGC_R_441K (0x5 << 5) +#define RT5639_DRC_AGC_R_882K (0x6 << 5) +#define RT5639_DRC_AGC_R_1764K (0x7 << 5) +#define RT5639_DRC_AGC_RC_MASK (0x1f) +#define RT5639_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5639_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5639_DRC_AGC_POB_SFT 8 +#define RT5639_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5639_DRC_AGC_CP_SFT 7 +#define RT5639_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5639_DRC_AGC_CP_EN (0x1 << 7) +#define RT5639_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5639_DRC_AGC_CPR_SFT 5 +#define RT5639_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5639_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5639_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5639_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5639_DRC_AGC_PRB_MASK (0x1f) +#define RT5639_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5639_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5639_DRC_AGC_NGB_SFT 12 +#define RT5639_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5639_DRC_AGC_TAR_SFT 7 +#define RT5639_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5639_DRC_AGC_NG_SFT 6 +#define RT5639_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5639_DRC_AGC_NG_EN (0x1 << 6) +#define RT5639_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5639_DRC_AGC_NGH_SFT 5 +#define RT5639_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5639_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5639_DRC_AGC_NGT_MASK (0x1f) +#define RT5639_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5639_JD_MASK (0x7 << 13) +#define RT5639_JD_SFT 13 +#define RT5639_JD_DIS (0x0 << 13) +#define RT5639_JD_GPIO1 (0x1 << 13) +#define RT5639_JD_JD1_IN4P (0x2 << 13) +#define RT5639_JD_JD2_IN4N (0x3 << 13) +#define RT5639_JD_GPIO2 (0x4 << 13) +#define RT5639_JD_GPIO3 (0x5 << 13) +#define RT5639_JD_GPIO4 (0x6 << 13) +#define RT5639_JD_HP_MASK (0x1 << 11) +#define RT5639_JD_HP_SFT 11 +#define RT5639_JD_HP_DIS (0x0 << 11) +#define RT5639_JD_HP_EN (0x1 << 11) +#define RT5639_JD_HP_TRG_MASK (0x1 << 10) +#define RT5639_JD_HP_TRG_SFT 10 +#define RT5639_JD_HP_TRG_LO (0x0 << 10) +#define RT5639_JD_HP_TRG_HI (0x1 << 10) +#define RT5639_JD_SPL_MASK (0x1 << 9) +#define RT5639_JD_SPL_SFT 9 +#define RT5639_JD_SPL_DIS (0x0 << 9) +#define RT5639_JD_SPL_EN (0x1 << 9) +#define RT5639_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5639_JD_SPL_TRG_SFT 8 +#define RT5639_JD_SPL_TRG_LO (0x0 << 8) +#define RT5639_JD_SPL_TRG_HI (0x1 << 8) +#define RT5639_JD_SPR_MASK (0x1 << 7) +#define RT5639_JD_SPR_SFT 7 +#define RT5639_JD_SPR_DIS (0x0 << 7) +#define RT5639_JD_SPR_EN (0x1 << 7) +#define RT5639_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5639_JD_SPR_TRG_SFT 6 +#define RT5639_JD_SPR_TRG_LO (0x0 << 6) +#define RT5639_JD_SPR_TRG_HI (0x1 << 6) +#define RT5639_JD_MO_MASK (0x1 << 5) +#define RT5639_JD_MO_SFT 5 +#define RT5639_JD_MO_DIS (0x0 << 5) +#define RT5639_JD_MO_EN (0x1 << 5) +#define RT5639_JD_MO_TRG_MASK (0x1 << 4) +#define RT5639_JD_MO_TRG_SFT 4 +#define RT5639_JD_MO_TRG_LO (0x0 << 4) +#define RT5639_JD_MO_TRG_HI (0x1 << 4) +#define RT5639_JD_LO_MASK (0x1 << 3) +#define RT5639_JD_LO_SFT 3 +#define RT5639_JD_LO_DIS (0x0 << 3) +#define RT5639_JD_LO_EN (0x1 << 3) +#define RT5639_JD_LO_TRG_MASK (0x1 << 2) +#define RT5639_JD_LO_TRG_SFT 2 +#define RT5639_JD_LO_TRG_LO (0x0 << 2) +#define RT5639_JD_LO_TRG_HI (0x1 << 2) +#define RT5639_JD1_IN4P_MASK (0x1 << 1) +#define RT5639_JD1_IN4P_SFT 1 +#define RT5639_JD1_IN4P_DIS (0x0 << 1) +#define RT5639_JD1_IN4P_EN (0x1 << 1) +#define RT5639_JD2_IN4N_MASK (0x1) +#define RT5639_JD2_IN4N_SFT 0 +#define RT5639_JD2_IN4N_DIS (0x0) +#define RT5639_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5639_IRQ_JD_MASK (0x1 << 15) +#define RT5639_IRQ_JD_SFT 15 +#define RT5639_IRQ_JD_BP (0x0 << 15) +#define RT5639_IRQ_JD_NOR (0x1 << 15) +#define RT5639_IRQ_OT_MASK (0x1 << 14) +#define RT5639_IRQ_OT_SFT 14 +#define RT5639_IRQ_OT_BP (0x0 << 14) +#define RT5639_IRQ_OT_NOR (0x1 << 14) +#define RT5639_JD_STKY_MASK (0x1 << 13) +#define RT5639_JD_STKY_SFT 13 +#define RT5639_JD_STKY_DIS (0x0 << 13) +#define RT5639_JD_STKY_EN (0x1 << 13) +#define RT5639_OT_STKY_MASK (0x1 << 12) +#define RT5639_OT_STKY_SFT 12 +#define RT5639_OT_STKY_DIS (0x0 << 12) +#define RT5639_OT_STKY_EN (0x1 << 12) +#define RT5639_JD_P_MASK (0x1 << 11) +#define RT5639_JD_P_SFT 11 +#define RT5639_JD_P_NOR (0x0 << 11) +#define RT5639_JD_P_INV (0x1 << 11) +#define RT5639_OT_P_MASK (0x1 << 10) +#define RT5639_OT_P_SFT 10 +#define RT5639_OT_P_NOR (0x0 << 10) +#define RT5639_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5639_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5639_IRQ_MB1_OC_SFT 15 +#define RT5639_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5639_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5639_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5639_IRQ_MB2_OC_SFT 14 +#define RT5639_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5639_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5639_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5639_MB1_OC_STKY_SFT 11 +#define RT5639_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5639_MB1_OC_STKY_EN (0x1 << 11) +#define RT5639_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5639_MB2_OC_STKY_SFT 10 +#define RT5639_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5639_MB2_OC_STKY_EN (0x1 << 10) +#define RT5639_MB1_OC_P_MASK (0x1 << 7) +#define RT5639_MB1_OC_P_SFT 7 +#define RT5639_MB1_OC_P_NOR (0x0 << 7) +#define RT5639_MB1_OC_P_INV (0x1 << 7) +#define RT5639_MB2_OC_P_MASK (0x1 << 6) +#define RT5639_MB2_OC_P_SFT 6 +#define RT5639_MB2_OC_P_NOR (0x0 << 6) +#define RT5639_MB2_OC_P_INV (0x1 << 6) +#define RT5639_MB1_OC_CLR (0x1 << 3) +#define RT5639_MB1_OC_CLR_SFT 3 +#define RT5639_MB2_OC_CLR (0x1 << 2) +#define RT5639_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5639_GP1_PIN_MASK (0x1 << 15) +#define RT5639_GP1_PIN_SFT 15 +#define RT5639_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5639_GP1_PIN_IRQ (0x1 << 15) +#define RT5639_GP2_PIN_MASK (0x1 << 14) +#define RT5639_GP2_PIN_SFT 14 +#define RT5639_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5639_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5639_GP3_PIN_MASK (0x3 << 12) +#define RT5639_GP3_PIN_SFT 12 +#define RT5639_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5639_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5639_GP3_PIN_IRQ (0x2 << 12) +#define RT5639_GP4_PIN_MASK (0x1 << 11) +#define RT5639_GP4_PIN_SFT 11 +#define RT5639_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5639_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5639_DP_SIG_MASK (0x1 << 10) +#define RT5639_DP_SIG_SFT 10 +#define RT5639_DP_SIG_TEST (0x0 << 10) +#define RT5639_DP_SIG_AP (0x1 << 10) +#define RT5639_GPIO_M_MASK (0x1 << 9) +#define RT5639_GPIO_M_SFT 9 +#define RT5639_GPIO_M_FLT (0x0 << 9) +#define RT5639_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5639_GP4_PF_MASK (0x1 << 11) +#define RT5639_GP4_PF_SFT 11 +#define RT5639_GP4_PF_IN (0x0 << 11) +#define RT5639_GP4_PF_OUT (0x1 << 11) +#define RT5639_GP4_OUT_MASK (0x1 << 10) +#define RT5639_GP4_OUT_SFT 10 +#define RT5639_GP4_OUT_LO (0x0 << 10) +#define RT5639_GP4_OUT_HI (0x1 << 10) +#define RT5639_GP4_P_MASK (0x1 << 9) +#define RT5639_GP4_P_SFT 9 +#define RT5639_GP4_P_NOR (0x0 << 9) +#define RT5639_GP4_P_INV (0x1 << 9) +#define RT5639_GP3_PF_MASK (0x1 << 8) +#define RT5639_GP3_PF_SFT 8 +#define RT5639_GP3_PF_IN (0x0 << 8) +#define RT5639_GP3_PF_OUT (0x1 << 8) +#define RT5639_GP3_OUT_MASK (0x1 << 7) +#define RT5639_GP3_OUT_SFT 7 +#define RT5639_GP3_OUT_LO (0x0 << 7) +#define RT5639_GP3_OUT_HI (0x1 << 7) +#define RT5639_GP3_P_MASK (0x1 << 6) +#define RT5639_GP3_P_SFT 6 +#define RT5639_GP3_P_NOR (0x0 << 6) +#define RT5639_GP3_P_INV (0x1 << 6) +#define RT5639_GP2_PF_MASK (0x1 << 5) +#define RT5639_GP2_PF_SFT 5 +#define RT5639_GP2_PF_IN (0x0 << 5) +#define RT5639_GP2_PF_OUT (0x1 << 5) +#define RT5639_GP2_OUT_MASK (0x1 << 4) +#define RT5639_GP2_OUT_SFT 4 +#define RT5639_GP2_OUT_LO (0x0 << 4) +#define RT5639_GP2_OUT_HI (0x1 << 4) +#define RT5639_GP2_P_MASK (0x1 << 3) +#define RT5639_GP2_P_SFT 3 +#define RT5639_GP2_P_NOR (0x0 << 3) +#define RT5639_GP2_P_INV (0x1 << 3) +#define RT5639_GP1_PF_MASK (0x1 << 2) +#define RT5639_GP1_PF_SFT 2 +#define RT5639_GP1_PF_IN (0x0 << 2) +#define RT5639_GP1_PF_OUT (0x1 << 2) +#define RT5639_GP1_OUT_MASK (0x1 << 1) +#define RT5639_GP1_OUT_SFT 1 +#define RT5639_GP1_OUT_LO (0x0 << 1) +#define RT5639_GP1_OUT_HI (0x1 << 1) +#define RT5639_GP1_P_MASK (0x1) +#define RT5639_GP1_P_SFT 0 +#define RT5639_GP1_P_NOR (0x0) +#define RT5639_GP1_P_INV (0x1) + +/* FM34-500 Register Control 1 (0xc4) */ +#define RT5639_DSP_ADD_SFT 0 + +/* FM34-500 Register Control 2 (0xc5) */ +#define RT5639_DSP_DAT_SFT 0 + +/* FM34-500 Register Control 3 (0xc6) */ +#define RT5639_DSP_BUSY_MASK (0x1 << 15) +#define RT5639_DSP_BUSY_BIT 15 +#define RT5639_DSP_DS_MASK (0x1 << 14) +#define RT5639_DSP_DS_SFT 14 +#define RT5639_DSP_DS_FM3010 (0x1 << 14) +#define RT5639_DSP_DS_TEMP (0x1 << 14) +#define RT5639_DSP_CLK_MASK (0x3 << 12) +#define RT5639_DSP_CLK_SFT 12 +#define RT5639_DSP_CLK_384K (0x0 << 12) +#define RT5639_DSP_CLK_192K (0x1 << 12) +#define RT5639_DSP_CLK_96K (0x2 << 12) +#define RT5639_DSP_CLK_64K (0x3 << 12) +#define RT5639_DSP_PD_PIN_MASK (0x1 << 11) +#define RT5639_DSP_PD_PIN_SFT 11 +#define RT5639_DSP_PD_PIN_LO (0x0 << 11) +#define RT5639_DSP_PD_PIN_HI (0x1 << 11) +#define RT5639_DSP_RST_PIN_MASK (0x1 << 10) +#define RT5639_DSP_RST_PIN_SFT 10 +#define RT5639_DSP_RST_PIN_LO (0x0 << 10) +#define RT5639_DSP_RST_PIN_HI (0x1 << 10) +#define RT5639_DSP_R_EN (0x1 << 9) +#define RT5639_DSP_R_EN_BIT 9 +#define RT5639_DSP_W_EN (0x1 << 8) +#define RT5639_DSP_W_EN_BIT 8 +#define RT5639_DSP_CMD_MASK (0xff) +#define RT5639_DSP_CMD_SFT 0 +#define RT5639_DSP_CMD_MW (0x3B) /* Memory Write */ +#define RT5639_DSP_CMD_MR (0x37) /* Memory Read */ +#define RT5639_DSP_CMD_RR (0x60) /* Register Read */ +#define RT5639_DSP_CMD_RW (0x68) /* Register Write */ + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5639_REG_SEQ_MASK (0xf << 12) +#define RT5639_REG_SEQ_SFT 12 +#define RT5639_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5639_SEQ1_ST_SFT 11 +#define RT5639_SEQ1_ST_RUN (0x0 << 11) +#define RT5639_SEQ1_ST_FIN (0x1 << 11) +#define RT5639_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5639_SEQ2_ST_SFT 10 +#define RT5639_SEQ2_ST_RUN (0x0 << 10) +#define RT5639_SEQ2_ST_FIN (0x1 << 10) +#define RT5639_REG_LV_MASK (0x1 << 9) +#define RT5639_REG_LV_SFT 9 +#define RT5639_REG_LV_MX (0x0 << 9) +#define RT5639_REG_LV_PR (0x1 << 9) +#define RT5639_SEQ_2_PT_MASK (0x1 << 8) +#define RT5639_SEQ_2_PT_BIT 8 +#define RT5639_REG_IDX_MASK (0xff) +#define RT5639_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5639_REG_DAT_MASK (0xffff) +#define RT5639_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5639_SEQ_DLY_MASK (0xff << 8) +#define RT5639_SEQ_DLY_SFT 8 +#define RT5639_PROG_MASK (0x1 << 7) +#define RT5639_PROG_SFT 7 +#define RT5639_PROG_DIS (0x0 << 7) +#define RT5639_PROG_EN (0x1 << 7) +#define RT5639_SEQ1_PT_RUN (0x1 << 6) +#define RT5639_SEQ1_PT_RUN_BIT 6 +#define RT5639_SEQ2_PT_RUN (0x1 << 5) +#define RT5639_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5639_SEQ1_START_MASK (0xf << 8) +#define RT5639_SEQ1_START_SFT 8 +#define RT5639_SEQ1_END_MASK (0xf) +#define RT5639_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5639_SEQ2_START_MASK (0xf << 8) +#define RT5639_SEQ2_START_SFT 8 +#define RT5639_SEQ2_END_MASK (0xf) +#define RT5639_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5639_SCB_KEY_MASK (0xff) +#define RT5639_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5639_SCB_SWAP_MASK (0x1 << 15) +#define RT5639_SCB_SWAP_SFT 15 +#define RT5639_SCB_SWAP_DIS (0x0 << 15) +#define RT5639_SCB_SWAP_EN (0x1 << 15) +#define RT5639_SCB_MASK (0x1 << 14) +#define RT5639_SCB_SFT 14 +#define RT5639_SCB_DIS (0x0 << 14) +#define RT5639_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5639_BB_MASK (0x1 << 15) +#define RT5639_BB_SFT 15 +#define RT5639_BB_DIS (0x0 << 15) +#define RT5639_BB_EN (0x1 << 15) +#define RT5639_BB_CT_MASK (0x7 << 12) +#define RT5639_BB_CT_SFT 12 +#define RT5639_BB_CT_A (0x0 << 12) +#define RT5639_BB_CT_B (0x1 << 12) +#define RT5639_BB_CT_C (0x2 << 12) +#define RT5639_BB_CT_D (0x3 << 12) +#define RT5639_M_BB_L_MASK (0x1 << 9) +#define RT5639_M_BB_L_SFT 9 +#define RT5639_M_BB_R_MASK (0x1 << 8) +#define RT5639_M_BB_R_SFT 8 +#define RT5639_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5639_M_BB_HPF_L_SFT 7 +#define RT5639_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5639_M_BB_HPF_R_SFT 6 +#define RT5639_G_BB_BST_MASK (0x3f) +#define RT5639_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5639_M_MP3_L_MASK (0x1 << 15) +#define RT5639_M_MP3_L_SFT 15 +#define RT5639_M_MP3_R_MASK (0x1 << 14) +#define RT5639_M_MP3_R_SFT 14 +#define RT5639_M_MP3_MASK (0x1 << 13) +#define RT5639_M_MP3_SFT 13 +#define RT5639_M_MP3_DIS (0x0 << 13) +#define RT5639_M_MP3_EN (0x1 << 13) +#define RT5639_EG_MP3_MASK (0x1f << 8) +#define RT5639_EG_MP3_SFT 8 +#define RT5639_MP3_HLP_MASK (0x1 << 7) +#define RT5639_MP3_HLP_SFT 7 +#define RT5639_MP3_HLP_DIS (0x0 << 7) +#define RT5639_MP3_HLP_EN (0x1 << 7) +#define RT5639_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5639_M_MP3_ORG_L_SFT 6 +#define RT5639_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5639_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5639_MP3_WT_MASK (0x1 << 13) +#define RT5639_MP3_WT_SFT 13 +#define RT5639_MP3_WT_1_4 (0x0 << 13) +#define RT5639_MP3_WT_1_2 (0x1 << 13) +#define RT5639_OG_MP3_MASK (0x1f << 8) +#define RT5639_OG_MP3_SFT 8 +#define RT5639_HG_MP3_MASK (0x3f) +#define RT5639_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5639_3D_CF_MASK (0x1 << 15) +#define RT5639_3D_CF_SFT 15 +#define RT5639_3D_CF_DIS (0x0 << 15) +#define RT5639_3D_CF_EN (0x1 << 15) +#define RT5639_3D_HP_MASK (0x1 << 14) +#define RT5639_3D_HP_SFT 14 +#define RT5639_3D_HP_DIS (0x0 << 14) +#define RT5639_3D_HP_EN (0x1 << 14) +#define RT5639_3D_BT_MASK (0x1 << 13) +#define RT5639_3D_BT_SFT 13 +#define RT5639_3D_BT_DIS (0x0 << 13) +#define RT5639_3D_BT_EN (0x1 << 13) +#define RT5639_3D_1F_MIX_MASK (0x3 << 11) +#define RT5639_3D_1F_MIX_SFT 11 +#define RT5639_3D_HP_M_MASK (0x1 << 10) +#define RT5639_3D_HP_M_SFT 10 +#define RT5639_3D_HP_M_SUR (0x0 << 10) +#define RT5639_3D_HP_M_FRO (0x1 << 10) +#define RT5639_M_3D_HRTF_MASK (0x1 << 9) +#define RT5639_M_3D_HRTF_SFT 9 +#define RT5639_M_3D_D2H_MASK (0x1 << 8) +#define RT5639_M_3D_D2H_SFT 8 +#define RT5639_M_3D_D2R_MASK (0x1 << 7) +#define RT5639_M_3D_D2R_SFT 7 +#define RT5639_M_3D_REVB_MASK (0x1 << 6) +#define RT5639_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5639_2ND_HPF_MASK (0x1 << 15) +#define RT5639_2ND_HPF_SFT 15 +#define RT5639_2ND_HPF_DIS (0x0 << 15) +#define RT5639_2ND_HPF_EN (0x1 << 15) +#define RT5639_HPF_CF_L_MASK (0x7 << 12) +#define RT5639_HPF_CF_L_SFT 12 +#define RT5639_1ST_HPF_MASK (0x1 << 11) +#define RT5639_1ST_HPF_SFT 11 +#define RT5639_1ST_HPF_DIS (0x0 << 11) +#define RT5639_1ST_HPF_EN (0x1 << 11) +#define RT5639_HPF_CF_R_MASK (0x7 << 8) +#define RT5639_HPF_CF_R_SFT 8 +#define RT5639_ZD_T_MASK (0x3 << 6) +#define RT5639_ZD_T_SFT 6 +#define RT5639_ZD_F_MASK (0x3 << 4) +#define RT5639_ZD_F_SFT 4 +#define RT5639_ZD_F_IM (0x0 << 4) +#define RT5639_ZD_F_ZC_IM (0x1 << 4) +#define RT5639_ZD_F_ZC_IOD (0x2 << 4) +#define RT5639_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5639_SI_DAC_MASK (0x1 << 11) +#define RT5639_SI_DAC_SFT 11 +#define RT5639_SI_DAC_AUTO (0x0 << 11) +#define RT5639_SI_DAC_TEST (0x1 << 11) +#define RT5639_DC_CAL_M_MASK (0x1 << 10) +#define RT5639_DC_CAL_M_SFT 10 +#define RT5639_DC_CAL_M_CAL (0x0 << 10) +#define RT5639_DC_CAL_M_NOR (0x1 << 10) +#define RT5639_DC_CAL_MASK (0x1 << 9) +#define RT5639_DC_CAL_SFT 9 +#define RT5639_DC_CAL_DIS (0x0 << 9) +#define RT5639_DC_CAL_EN (0x1 << 9) +#define RT5639_HPD_RCV_MASK (0x7 << 6) +#define RT5639_HPD_RCV_SFT 6 +#define RT5639_HPD_PS_MASK (0x1 << 5) +#define RT5639_HPD_PS_SFT 5 +#define RT5639_HPD_PS_DIS (0x0 << 5) +#define RT5639_HPD_PS_EN (0x1 << 5) +#define RT5639_CAL_M_MASK (0x1 << 4) +#define RT5639_CAL_M_SFT 4 +#define RT5639_CAL_M_DEP (0x0 << 4) +#define RT5639_CAL_M_CAL (0x1 << 4) +#define RT5639_CAL_MASK (0x1 << 3) +#define RT5639_CAL_SFT 3 +#define RT5639_CAL_DIS (0x0 << 3) +#define RT5639_CAL_EN (0x1 << 3) +#define RT5639_CAL_TEST_MASK (0x1 << 2) +#define RT5639_CAL_TEST_SFT 2 +#define RT5639_CAL_TEST_DIS (0x0 << 2) +#define RT5639_CAL_TEST_EN (0x1 << 2) +#define RT5639_CAL_P_MASK (0x3) +#define RT5639_CAL_P_SFT 0 +#define RT5639_CAL_P_NONE (0x0) +#define RT5639_CAL_P_CAL (0x1) +#define RT5639_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5639_SV_MASK (0x1 << 15) +#define RT5639_SV_SFT 15 +#define RT5639_SV_DIS (0x0 << 15) +#define RT5639_SV_EN (0x1 << 15) +#define RT5639_SPO_SV_MASK (0x1 << 14) +#define RT5639_SPO_SV_SFT 14 +#define RT5639_SPO_SV_DIS (0x0 << 14) +#define RT5639_SPO_SV_EN (0x1 << 14) +#define RT5639_OUT_SV_MASK (0x1 << 13) +#define RT5639_OUT_SV_SFT 13 +#define RT5639_OUT_SV_DIS (0x0 << 13) +#define RT5639_OUT_SV_EN (0x1 << 13) +#define RT5639_HP_SV_MASK (0x1 << 12) +#define RT5639_HP_SV_SFT 12 +#define RT5639_HP_SV_DIS (0x0 << 12) +#define RT5639_HP_SV_EN (0x1 << 12) +#define RT5639_ZCD_DIG_MASK (0x1 << 11) +#define RT5639_ZCD_DIG_SFT 11 +#define RT5639_ZCD_DIG_DIS (0x0 << 11) +#define RT5639_ZCD_DIG_EN (0x1 << 11) +#define RT5639_ZCD_MASK (0x1 << 10) +#define RT5639_ZCD_SFT 10 +#define RT5639_ZCD_PD (0x0 << 10) +#define RT5639_ZCD_PU (0x1 << 10) +#define RT5639_M_ZCD_MASK (0x3f << 4) +#define RT5639_M_ZCD_SFT 4 +#define RT5639_M_ZCD_RM_L (0x1 << 9) +#define RT5639_M_ZCD_RM_R (0x1 << 8) +#define RT5639_M_ZCD_SM_L (0x1 << 7) +#define RT5639_M_ZCD_SM_R (0x1 << 6) +#define RT5639_M_ZCD_OM_L (0x1 << 5) +#define RT5639_M_ZCD_OM_R (0x1 << 4) +#define RT5639_SV_DLY_MASK (0xf) +#define RT5639_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5639_ZCD_HP_MASK (0x1 << 15) +#define RT5639_ZCD_HP_SFT 15 +#define RT5639_ZCD_HP_DIS (0x0 << 15) +#define RT5639_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5639_3D_SPK_MASK (0x1 << 15) +#define RT5639_3D_SPK_SFT 15 +#define RT5639_3D_SPK_DIS (0x0 << 15) +#define RT5639_3D_SPK_EN (0x1 << 15) +#define RT5639_3D_SPK_M_MASK (0x3 << 13) +#define RT5639_3D_SPK_M_SFT 13 +#define RT5639_3D_SPK_CG_MASK (0x1f << 8) +#define RT5639_3D_SPK_CG_SFT 8 +#define RT5639_3D_SPK_SG_MASK (0x1f) +#define RT5639_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5639_WND_MASK (0x1 << 15) +#define RT5639_WND_SFT 15 +#define RT5639_WND_DIS (0x0 << 15) +#define RT5639_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5639_WND_FC_NW_MASK (0x3f << 10) +#define RT5639_WND_FC_NW_SFT 10 +#define RT5639_WND_FC_WK_MASK (0x3f << 4) +#define RT5639_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5639_HPF_FC_MASK (0x3f << 6) +#define RT5639_HPF_FC_SFT 6 +#define RT5639_WND_FC_ST_MASK (0x3f) +#define RT5639_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5639_WND_TH_LO_MASK (0x3ff) +#define RT5639_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5639_WND_TH_HI_MASK (0x3ff) +#define RT5639_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5639_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5639_WND_WIND_SFT 13 +#define RT5639_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5639_WND_STRONG_SFT 12 +enum { + RT5639_NO_WIND, + RT5639_BREEZE, + RT5639_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5639_DP_ATT_MASK (0x3 << 14) +#define RT5639_DP_ATT_SFT 14 +#define RT5639_DP_SPK_MASK (0x1 << 10) +#define RT5639_DP_SPK_SFT 10 +#define RT5639_DP_SPK_DIS (0x0 << 10) +#define RT5639_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5639_EQ_PRE_VOL_MASK (0xffff) +#define RT5639_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5639_EQ_PST_VOL_MASK (0xffff) +#define RT5639_EQ_PST_VOL_SFT 0 + +#define RT5639_NO_JACK BIT(0) +#define RT5639_HEADSET_DET BIT(1) +#define RT5639_HEADPHO_DET BIT(2) + +/* System Clock Source */ +#define RT5639_SCLK_S_MCLK 0 +#define RT5639_SCLK_S_PLL1 1 +#define RT5639_SCLK_S_PLL1_TK 2 +#define RT5639_SCLK_S_RCCLK 3 + +/* PLL1 Source */ +#define RT5639_PLL1_S_MCLK 0 +#define RT5639_PLL1_S_BCLK1 1 +#define RT5639_PLL1_S_BCLK2 2 +#define RT5639_PLL1_S_BCLK3 3 + + +enum { + RT5639_AIF1, + RT5639_AIF2, + RT5639_AIF3, + RT5639_AIFS, +}; + +enum { + RT5639_U_IF1 = 0x1, + RT5639_U_IF2 = 0x2, + RT5639_U_IF3 = 0x4, +}; + +enum { + RT5639_IF_123, + RT5639_IF_132, + RT5639_IF_312, + RT5639_IF_321, + RT5639_IF_231, + RT5639_IF_213, + RT5639_IF_113, + RT5639_IF_223, + RT5639_IF_ALL, +}; + +enum { + RT5639_DMIC_DIS, + RT5639_DMIC1, + RT5639_DMIC2, +}; + +struct rt5639_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5639_priv { + struct snd_soc_codec *codec; + struct rt5639_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5639_AIFS]; + int bclk[RT5639_AIFS]; + int master[RT5639_AIFS]; + + struct rt5639_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + bool hp_mute; +}; + +#endif -- 1.8.1.1.439.g50a6b54

5 9

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for three Dell laptops
by Hui Wang 26 Mar '14

26 Mar '14

When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255, SID: 0x10280632; VID: 0x10ec0293, SID: 0x1028062c; VID: 0x10ec0293, SID: 0x1028062e), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b0e25e5..af5f903 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4586,6 +4586,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x062c, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x062e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0632, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [Audio Driver Low power state]
by Gomathi Kumar 26 Mar '14

26 Mar '14

Hi all, I am new to kernel work and am trying to understand the low power state of audio driver. I am using ALC262 codec, hda_intel and 3.10 kernel code. In this codec, d3_stop_clk flag is not set. So in the code flow I was able to understand the following. if (!codec->pm_down_notified && !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) this particular condition in hda_power_work function fails, as power_keep_link_on flag is directly based on d3_stop_clk flag. codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_CLKSTOP); I modified the if condition to not check the power_keep_link_on flag. After this change the audio is consuming lesser power, but I am not able to see it going to suspend/runtime suspend. *Please clarify what is happening when I modify the if condition and how audio device is consuming low power?* *Also please help me understand, the significance of flags EPSS and D3_STOP_CLK and if they can be set by software code. * Thanks, Gomathi

1 0

[alsa-devel] [RFC v2 PATCH] ASoC: wm8904: add CCF support
by Bo Shen 26 Mar '14

26 Mar '14

Enable WM8904 to support common clock framework. - Now it supports use SoC provided clock or external oscillator as MCLK. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Documentation/devicetree/bindings/sound/wm8904.txt | 48 ++++++++++++++++++++++ sound/soc/codecs/wm8904.c | 17 ++++++++ 2 files changed, 65 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 0000000..bcf4aa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,48 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + - clocks and clock-names: reference to + <Documentation/devicetree/bindings/clock/clock-bindings.txt> + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +1. Using SoC provided clock as mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; +}; + +2. Using external oscillator as mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + + wm8904_osc { + compatible = "fixed-clock"; + #clock-cells = <0>; + #clock-frequency = <12000000>; + } + + clocks = <&wm8904_osc>; + clock-names = "mclk"; +}; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 49c35c3..4db6a26 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -49,6 +50,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; enum wm8904_type devtype; @@ -1828,6 +1830,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (wm8904->mclk) + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1899,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (wm8904->mclk) + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2139,6 +2147,15 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + dev_err(&i2c->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(wm8904->mclk); + goto err_enable; + } + } + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); if (ret < 0) { dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); -- 1.8.5.2

4 4

[alsa-devel] [PATCH v3] ASOC: tegra: move AC97 clock handling to the machine driver
by Lucas Stach 25 Mar '14

25 Mar '14

On Tegra the convention is to have a single machine driver that's controlling the whole audio subsystem. This was introduced after the AC97 driver was merged and ever since AC97 has been broken. Move the clock handling to the machine driver, to be in line with the other Tegra drivers and give the machine driver full control over the single Tegra audio PLL. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> -- v2: - remove redundant clock rate setting - defer removal of err_clk_put label to avoid conflict with Stephens dma channel cleanup v3: - rebased to 3.14-rc1 - no functional change - reword commit message CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Thierry Reding <thierry.reding(a)gmail.com> CC: Stephen Warren <swarren(a)wwwdotorg.org> CC: Stefan Agner <stefan(a)agner.ch> CC: alsa-devel(a)alsa-project.org --- sound/soc/tegra/tegra20_ac97.c | 15 +-------------- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra_wm9712.c | 17 ++++++++++++++++- 3 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index cf5e1cf..29f8832 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -37,7 +37,6 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include "tegra_asoc_utils.h" #include "tegra20_ac97.h" #define DRV_NAME "tegra20-ac97" @@ -376,18 +375,10 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; ac97->playback_dma_data.maxburst = 4; - ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); - if (ret) - goto err_clk_put; - - ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); - if (ret) - goto err_asoc_utils_fini; - ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err_asoc_utils_fini; + goto err; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); @@ -419,8 +410,6 @@ err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_clk_disable_unprepare: clk_disable_unprepare(ac97->clk_ac97); -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: snd_soc_set_ac97_ops(NULL); @@ -434,8 +423,6 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - tegra_asoc_utils_fini(&ac97->util_data); - clk_disable_unprepare(ac97->clk_ac97); snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h index 4acb3aa..0a39d82 100644 --- a/sound/soc/tegra/tegra20_ac97.h +++ b/sound/soc/tegra/tegra20_ac97.h @@ -90,6 +90,5 @@ struct tegra20_ac97 { struct regmap *regmap; int reset_gpio; int sync_gpio; - struct tegra_asoc_utils_data util_data; }; #endif /* __TEGRA20_AC97_H__ */ diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 45b5789..25a7f82 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -29,10 +29,13 @@ #include <sound/pcm_params.h> #include <sound/soc.h> +#include "tegra_asoc_utils.h" + #define DRV_NAME "tegra-snd-wm9712" struct tegra_wm9712 { struct platform_device *codec; + struct tegra_asoc_utils_data util_data; }; static const struct snd_soc_dapm_widget tegra_wm9712_dapm_widgets[] = { @@ -118,15 +121,25 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) tegra_wm9712_dai.platform_of_node = tegra_wm9712_dai.cpu_of_node; + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto codec_unregister; + + ret = tegra_asoc_utils_set_ac97_rate(&machine->util_data); + if (ret) + goto asoc_utils_fini; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto codec_unregister; + goto asoc_utils_fini; } return 0; +asoc_utils_fini: + tegra_asoc_utils_fini(&machine->util_data); codec_unregister: platform_device_del(machine->codec); codec_put: @@ -141,6 +154,8 @@ static int tegra_wm9712_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); + tegra_asoc_utils_fini(&machine->util_data); + platform_device_unregister(machine->codec); return 0; -- 1.8.5.3

4 4

[alsa-devel] [PARTIAL PATCH 0/3] ASoC: simple-card: multi DAI links extension
by Jean-Francois Moine 25 Mar '14

25 Mar '14

This patch series extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. ** this is a partial patch ** Jyri, is this OK as a base for you? - partial patch - remove 'simple-card: Simplify code' which is applied (Mark Brown) - add a DAI link container in the DT - v3 - remove 'Fix the reference count of device nodes' which is applied (Mark Brown) - new patch 'Simplify code' - dynamically allocate and use properties for all DAI links (Jyri Sarha and Li Xiubo) - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (3): ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Add DT documentation for multi-DAI links ASoC: simple-card: Handle many DAI links .../devicetree/bindings/sound/simple-card.txt | 42 +++- sound/soc/generic/simple-card.c | 220 +++++++++++++-------- 2 files changed, 183 insertions(+), 79 deletions(-) -- 1.9.1

3 5

[alsa-devel] [RFC PATCH] alsa-lib: Call LADSPA activate after all ports are setup
by Matthias Larisch 25 Mar '14

25 Mar '14

Hey there, I notices mainly swh-plugins (like butterworth_1902) to crash on seek with mplayer or mpd. This is due to their behaviour on relying on the activate call after the control inputs are set. This is not right by LADSPA API description but I see no point in NOT calling activate as late as possible in alsa-lib. Comments? Regards, Matthias From f7f5d920c845fc54ffef208f68f568b06e8fadd1 Mon Sep 17 00:00:00 2001 From: Matthias Larisch <mail(a)matthias-larisch.de> Date: Mon, 17 Mar 2014 19:37:01 +0100 Subject: [PATCH] call ladspa activate as late as possible to work around wrong plugin behaviour --- src/pcm/pcm_ladspa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 9ce5242..7d1e3df 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -645,8 +645,6 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l return -EINVAL; } list_add_tail(&instance->list, &plugin->instances); - if (plugin->desc->activate) - plugin->desc->activate(instance->handle); if (plugin->policy == SND_PCM_LADSPA_POLICY_DUPLICATE) { err = snd_pcm_ladspa_connect_plugin_duplicate(plugin, &plugin->input, &plugin->output, instance, idx); if (err < 0) { @@ -664,6 +662,8 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l assert(err >= 0); err = snd_pcm_ladspa_connect_controls(plugin, &plugin->output, instance); assert(err >= 0); + if (plugin->desc->activate) + plugin->desc->activate(instance->handle); } err = snd_pcm_ladspa_check_connect(plugin, &plugin->input, &instance->input, depth); if (err < 0) -- 1.9.0

2 1

[alsa-devel] [PATCH 0/4] Add fsl_asoc_xlate_tdm_slot_mask()
by Xiubo Li 25 Mar '14

25 Mar '14

Xiubo Li (4): ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask. ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support. include/sound/soc-dai.h | 2 +- sound/soc/fsl/Kconfig | 2 ++ sound/soc/fsl/fsl_esai.c | 2 ++ sound/soc/fsl/fsl_utils.c | 27 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 4 +++- sound/soc/fsl/imx-ssi.c | 2 ++ sound/soc/soc-core.c | 10 +++++----- 7 files changed, 42 insertions(+), 7 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 0/19] ASoC: rsnd: new module path method and DT support
by Kuninori Morimoto 25 Mar '14

25 Mar '14

Hi Mark These are Renesas R-Car sound driver patches. #1 - #7 are clean up patches. #8 - #18 adds new module path method #19 adds DT support Now, Mark's ALSA branch doesn't include SH-ARM platform branch. Thus, these patches are keeping compatible for it at this point. Compatible code will be removed after merge. Kuninori Morimoto (19): 1) ASoC: rsnd: move priv member settings to upper side 2) ASoC: rsnd: move rsnd_mod_call() macro 3) ASoC: rsnd: remove verbose function parameter 4) ASoC: rsnd: remove verbose debug message from scu/ssi 5) ASoC: rsnd: unify rdai naming 6) ASoC: rsnd: add .set_sysclk on snd_soc_dai_ops 7) ASoC: rsnd: tidyup RSND_SSI_xxx flags 8) ASoC: rsnd: run rsnd_path_init() when probe() timing 9) ASoC: rsnd: use mod array instead of list on rdai 10) ASoC: rsnd: get ssi/scu from rsnd_dai_stream 11) ASoC: rsnd: add struct rsnd_dai_platform_info 12) ASoC: rsnd: use function pointer for each probe 13) ASoC: rsnd: Get correct SCU ID 14) ASoC: rsnd: add rsnd_scu_enable_ssi_irq() 15) ASoC: rsnd: call rsnd_scu_ssi_mode_init() from SSI 16) ASoC: rsnd: add probe/remove callback on rsnd_mod_ops 17) ASoC: rsnd: use mod probe method on SCU 18) ASoC: rsnd: use mod probe method on SSI 19) ASoC: rsnd: add DeviceTree support .../devicetree/bindings/sound/renesas,rsnd.txt | 101 ++++++ include/sound/rcar_snd.h | 21 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 380 ++++++++++++++++---- sound/soc/sh/rcar/gen.c | 78 +--- sound/soc/sh/rcar/rsnd.h | 86 +++-- sound/soc/sh/rcar/scu.c | 188 ++++++---- sound/soc/sh/rcar/ssi.c | 191 +++++++--- 8 files changed, 759 insertions(+), 288 deletions(-)

3 67

[alsa-devel] [PATCH v2 00/12] Cleanup card DAPM and control setup
by Lars-Peter Clausen 25 Mar '14

25 Mar '14

This patch series converts all remaining machine drivers that still register their controls and DAPM elements with the CODEC to registering them with the card. The reason for doing this is to be able to remove the hack in the ASoC core that un-sets the name_prefix of a CODEC before calling the init callback for a DAI link. This comes from a time where it was not possible to register controls or DAPM elements at a card level. And removing this hack makes going forward with the componentisation easier and it also looks like the DAI multi-CODEC patch could benefit from this as well. The series depends on topic/pxa and topic/dapm - Lars Lars-Peter Clausen (12): ASoC: ams-delta: Fix compile error ASoC: ams-delta: Convert to table based DAPM and control setup ASoC: mfld_machine: Convert to table based DAPM and control setup ASoC: omap-abe-twl6040: Register machine level DMIC DAPM routes with the card ASoC: omap: rx51: Convert to table based control and DAPM setup ASoC: omap3pandora: Convert to table based DAPM setup ASoC: davinci-evm: Register machine level DAPM elements with the card ASoC: snappercl15: Convert to table based DAPM setup ASoC: s6105-ipcam: Convert to table based DAPM setup ASoC: pxa: magician: Convert to table based DAPM and control setup ASoC: pxa: tosa: Convert to table based DAPM and control setup ASoC: Remove name_prefix un-set during DAI link init hack sound/soc/cirrus/snappercl15.c | 18 +++-------- sound/soc/davinci/davinci-evm.c | 22 +++++-------- sound/soc/intel/mfld_machine.c | 49 ++++++++++++----------------- sound/soc/omap/ams-delta.c | 66 +++++++++++++-------------------------- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/omap3pandora.c | 31 ++++++------------ sound/soc/omap/rx51.c | 47 +++++----------------------- sound/soc/pxa/magician.c | 34 +++++++------------- sound/soc/pxa/tosa.c | 35 ++++++++------------- sound/soc/s6000/s6105-ipcam.c | 28 ++++++----------- sound/soc/soc-core.c | 7 +---- 11 files changed, 108 insertions(+), 232 deletions(-) -- 1.8.0

3 25

[alsa-devel] [PATCH v3 0/3] ASoC: davinci: perparation for eDMA dmaengine PCM
by Peter Ujfalusi 25 Mar '14

25 Mar '14

Hi, Changes since v2: - Constraint only placed for the period size [1] - Patch for davinci-pcm.h to add empty function in case of !DAVINCI platforms When the edma arch and dmaengine patches has been taken only one patch is needed to switch AM335x/AM437x to use the edma-pcm. [1]. I was not able to reporoduce the issue I have seen with mplayer anymore. As it was thought constraint for the period size is enough and there's no need to place the same constraint on buffer size. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-mcasp: Constraint on the period and buffer size based on FIFO usage ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-pcm: Add empty functions for !CONFIG_SND_DAVINCI_SOC builds sound/soc/davinci/davinci-mcasp.c | 33 +++++++++++++++++++++++++++++---- sound/soc/davinci/davinci-pcm.h | 10 ++++++++++ 2 files changed, 39 insertions(+), 4 deletions(-) -- 1.9.0

3 13

[alsa-devel] [PATCH] ALSA: hda - Inform the unexpectedly ignored pins by auto-parser
by Takashi Iwai 25 Mar '14

25 Mar '14

The auto-parser may ignore some pins that could be valid when they don't match with the assumption or if there are way too many pins assigned to the same output type. So far, such a pin has been silently ignored, but it's better to leave a message, which would help for debugging and understanding the problem. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8de7cfa460b9..90d2fda6c8f9 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -227,10 +227,18 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; if (!assoc_line_out) assoc_line_out = assoc; - else if (assoc_line_out != assoc) + else if (assoc_line_out != assoc) { + codec_info(codec, + "ignore pin 0x%x with mismatching assoc# 0x%x vs 0x%x\n", + nid, assoc, assoc_line_out); continue; - if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) + } + if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } line_out[cfg->line_outs].pin = nid; line_out[cfg->line_outs].seq = seq; cfg->line_outs++; @@ -238,8 +246,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } speaker_out[cfg->speaker_outs].pin = nid; speaker_out[cfg->speaker_outs].seq = (assoc << 4) | seq; cfg->speaker_outs++; @@ -247,8 +259,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, case AC_JACK_HP_OUT: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } hp_out[cfg->hp_outs].pin = nid; hp_out[cfg->hp_outs].seq = (assoc << 4) | seq; cfg->hp_outs++; @@ -267,8 +283,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, break; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: - if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) { + codec_info(codec, + "ignore pin 0x%x, too many assigned pins\n", + nid); continue; + } cfg->dig_out_pins[cfg->dig_outs] = nid; cfg->dig_out_type[cfg->dig_outs] = (loc == AC_JACK_LOC_HDMI) ? -- 1.9.1

1 0

[alsa-devel] [RFC PATCH] ASoC: wm8904: add CCF support
by Bo Shen 25 Mar '14

25 Mar '14

Enable WM8904 to support common clock framework. And also supports different clk configuration. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Documentation/devicetree/bindings/sound/wm8904.txt | 57 ++++++++++++++++++++++ sound/soc/codecs/wm8904.c | 39 +++++++++++++++ 2 files changed, 96 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 0000000..039374d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,57 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + + if work with CCF, the following properties as required: + - wlf,sysclk-from-mclk: set the sys clock is driven from mclk, + - wlf,mclk-use-xtal: if the mclk is generated by crystal. + if without this property, the mclk is generated from SOC. + - clocks: which clock generated clock to mclk. + - clock-names: the clock names use for retrieve. + - wlf,mclk-freq: mclk's frequency + + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +1. General usage: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; +}; + +2. Working with CCF supports and using crystall provide mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + wlf,sysclk-from-mclk; + wlf,mclk-use-xtal; + wlf,mclk-freq = <12000000>; +}; + +3. Working with CCF supports and using SoC provide mclk: +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + wlf,sysclk-from-mclk; + clocks = <&pck0>; + clock-names = "mclk"; + wlf,mclk-freq = <32768>; +}; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 49c35c3..99e3e90 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -49,6 +50,10 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; + bool sysclk_from_mclk; + bool mclk_use_xtal; + u32 mclk_freq; enum wm8904_type devtype; @@ -1828,6 +1833,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (!wm8904->mclk_use_xtal) + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1902,9 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (!wm8904->mclk_use_xtal) + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2139,6 +2150,34 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + struct device_node *np = i2c->dev.of_node; + wm8904->sysclk_from_mclk = + of_property_read_bool(np, "wlf,sysclk-from-mclk"); + if (wm8904->sysclk_from_mclk) { + wm8904->mclk_use_xtal = + of_property_read_bool(np, "wlf,mclk-use-xtal"); + + ret = of_property_read_u32(np, "wlf,mclk-freq", + &wm8904->mclk_freq); + if (ret) { + dev_err(&i2c->dev, "Failed to read mclk freq"); + goto err_enable; + } + + if (!wm8904->mclk_use_xtal) { + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + dev_err(&i2c->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(wm8904->mclk); + goto err_enable; + } + + clk_set_rate(wm8904->mclk, wm8904->mclk_freq); + } + } + } + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); if (ret < 0) { dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); -- 1.8.5.2

4 8

[alsa-devel] [PATCH] ASoC: rcar: bugfix: it cares about the non-src case
by Kuninori Morimoto 24 Mar '14

24 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> src might not exist. kernel will be hung-up without this patch in such case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This is for mark/topic/rcar branch sound/soc/sh/rcar/src.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index eee75eb..6232b7d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -182,16 +182,20 @@ unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) { + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); struct rsnd_src *src; - unsigned int rate; + unsigned int rate = 0; - src = rsnd_mod_to_src(rsnd_io_to_mod_src(io)); + if (src_mod) { + src = rsnd_mod_to_src(src_mod); + + /* + * return convert rate if SRC is used, + * otherwise, return runtime->rate as usual + */ + rate = rsnd_src_convert_rate(src); + } - /* - * return convert rate if SRC is used, - * otherwise, return runtime->rate as usual - */ - rate = rsnd_src_convert_rate(src); if (!rate) rate = runtime->rate; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags
by Kuninori Morimoto 24 Mar '14

24 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SND_SOC_DAIFMT_CBx_CFx means "codec" side master/slave mode. Then, rcar will be master mode if it was SND_SOC_DAIFMT_CBS_CFS. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark, Simon I'm so sorry. current Mark's fix/rcar branch has SND_SOC_DAIFMT_CBx_CFx fixup patches, but it didn't care about bockw board. This patch is for it. This Subject is using "ARM: bockw:" prefix, but, should it be "ASoC: rcar" ? arch/arm/mach-shmobile/board-bockw.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/arch/arm/mach-shmobile/board-bockw.c b/arch/arm/mach-shmobile/board-bockw.c index b5867b7..f444be2 100644 --- a/arch/arm/mach-shmobile/board-bockw.c +++ b/arch/arm/mach-shmobile/board-bockw.c @@ -467,14 +467,12 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI56-AK4643", .codec = "ak4642-codec.0-0012", .platform = "rcar_sound", - .daifmt = SND_SOC_DAIFMT_LEFT_J, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "rsnd-dai.0", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "ak4642-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, .sysclk = 11289600, }, }, @@ -484,10 +482,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI3-AK4554(playback)", .codec = "ak4554-adc-dac.0", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_RIGHT_J, .cpu_dai = { .name = "rsnd-dai.1", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_RIGHT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -499,10 +496,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI4-AK4554(capture)", .codec = "ak4554-adc-dac.0", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_LEFT_J, .cpu_dai = { .name = "rsnd-dai.2", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_LEFT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -514,10 +510,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI7-AK4554(playback)", .codec = "ak4554-adc-dac.1", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_RIGHT_J, .cpu_dai = { .name = "rsnd-dai.3", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_RIGHT_J, }, .codec_dai = { .name = "ak4554-hifi", @@ -529,10 +524,9 @@ static struct asoc_simple_card_info rsnd_card_info[] = { .card = "SSI8-AK4554(capture)", .codec = "ak4554-adc-dac.1", .platform = "rcar_sound", + .daifmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_LEFT_J, .cpu_dai = { .name = "rsnd-dai.4", - .fmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_LEFT_J, }, .codec_dai = { .name = "ak4554-hifi", -- 1.7.9.5

2 2

[alsa-devel] [PATCH RFC 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 24 Mar '14

24 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. The patches implement the main part of the simple-card changes discussed in alsa-devel mailing list. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 91 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 237 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 201 insertions(+), 139 deletions(-) -- 1.7.9.5

3 6

[alsa-devel] [PATCH RFC v2 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 24 Mar '14

24 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. These patches implement the main part of the simple-card changes discussed in alsa-devel mailing list [2]. Since RFC version of the patches: - ASoC: core: Update snd_soc_of_parse_daifmt() interface - Make minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). (Kuninori Morimoto's comment [3]). - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Fix typo in DT binding document (Jean-Francois Moine's comment [4]). - Remove #define DEBUG - Remove accidental line wrap at simple-audio-card,name property descripition in the DT binding document [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074388.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074699.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074697.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 88 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 236 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 198 insertions(+), 138 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [hsw dp audio] Muting S/PDIF turns off DP
by Daniel Martin 24 Mar '14

24 Mar '14

Hi, I've a bug here: when playing audio via the speakers of a monitor attached to a DisplayPort and I mute the corresponding S/PDIF (i.e. with alsamixer) then the monitor switches off into standby, unmuting the S/PDIF turns the monitor back on. When muting the S/PDIF while no audio is playing everything is fine. There's no audio daemon running and my ~/.asoundrc contains only: pcm.!default { type hw card 0 device 3 } I've already filed a fdo bug report (drm/intel) for this as I thought it might be related to it: Bug 75729 - [hsw dp audio] Muting S/PDIF turns off DP https://bugs.freedesktop.org/show_bug.cgi?id=75729 Attached to this bug is a `alsa-info.log` too. As stated in the bug report it's not a showstopper for me, but obviously something seems to go wrong and maybe someone has an idea. Thanks, Daniel Martin

2 1

[alsa-devel] [GIT PULL] More ASoC updates for v3.15
by Mark Brown 24 Mar '14

24 Mar '14

The following changes since commit 9296f4da3bafa23d8b9abc5cd271a66ea8f90cd2: ASoC: tlv320aic31xx: Staticise non-exported symbols (2014-03-13 17:44:22 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-3 for you to fetch changes up to e090d5b6ad20056ec0ef58727e3ae95fd82be090: Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next (2014-03-23 14:00:59 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 A few more updates for the merge window: - Fixes for the simple-card DAI format DT mess. - A new driver for Cirrus cs42xx8 devices. - DT support for a couple more devices. - A revert of a previous buggy fix for soc-pcm, plus a few more fixes and cleanups. ---------------------------------------------------------------- Al Viro (3): ocfs2 syncs the wrong range... sockfd_lookup_light(): switch to fdget^W^Waway from fget_light get rid of fget_light() Ales Novak (1): [SCSI] storvsc: NULL pointer dereference fix Alex Deucher (4): drm/radeon: fix runpm disabling on non-PX harder drm/radeon/cik: properly set sdma ring status on disable drm/radeon/cik: stop the sdma engines in the enable() function drm/radeon/cik: properly set compute ring status on disable Alexander Aring (1): at86rf230: fix lockdep splats Alexei Starovoitov (1): x86: bpf_jit: support negative offsets Amir Vadai (2): net/mlx4_core: Fix memory access error in mlx4_QUERY_DEV_CAP_wrapper() net/mlx4_core: mlx4_init_slave() shouldn't access comm channel before PF is ready Amitkumar Karwar (2): mwifiex: copy AP's HT capability info correctly mwifiex: save and copy AP's VHT capability info correctly Andrew Lutomirski (1): net: Improve SO_TIMESTAMPING documentation and fix a minor code bug Andrew Morton (1): revert "kallsyms: fix absolute addresses for kASLR" Annie Li (1): Xen-netback: Fix issue caused by using gso_type wrongly Anton Blanchard (2): net: unix socket code abuses csum_partial ibmveth: Fix endian issues with MAC addresses Anton Nayshtut (1): ipv6: Fix exthdrs offload registration. Arnd Bergmann (1): vmxnet3: fix building without CONFIG_PCI_MSI Artem Fetishev (1): fs/proc/base.c: fix GPF in /proc/$PID/map_files Axel Lin (1): ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc Ben Hutchings (3): perf trace: Decode architecture-specific signal numbers bna: Replace large udelay() with mdelay() mm/Kconfig: fix URL for zsmalloc benchmark Bjorn Helgaas (3): phy: fix compiler array bounds warning on settings[] PCI: Enable INTx in pci_reenable_device() only when MSI/MSI-X not enabled PCI: Don't check resource_size() in pci_bus_alloc_resource() Brian Austin (3): ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols Chad Dupuis (1): [SCSI] qla2xxx: Fix multiqueue MSI-X registration. Colin Ian King (1): tools/testing/selftests/ipc/msgque.c: handle msgget failure return correctly Dan Williams (1): [SCSI] isci: fix reset timeout handling Daniel Borkmann (2): net: sctp: fix skb leakage in COOKIE ECHO path of chunk->auth_chunk MAINTAINERS: add networking selftests to NETWORKING Daniel J Blueman (1): x86/amd/numa: Fix northbridge quirk to assign correct NUMA node Dave Airlie (3): Merge tag 'vmwgfx-fixes-3.14-2014-03-13' of git://people.freedesktop.org/~thomash/linux into drm-fixes Merge branch 'drm-fixes-3.14' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge tag 'ttm-fixes-3.14-2014-03-12' of git://people.freedesktop.org/~thomash/linux into drm-fixes Dave Jones (2): perf/x86: Fix leak in uncore_type_init failure paths x86: Remove CONFIG_X86_OOSTORE David S. Miller (5): Merge branch 'tipc' Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'stmmac' Merge branch 'skb_frags' Merge branch 'mlx4' Don Zickus (1): perf machine: Use map as success in ip__resolve_ams Dylan Reid (1): ASoC: tegra: Use flat regcache Eliad Peller (1): mac80211: consider virtual mon when calculating min_def Emmanuel Grumbach (1): iwlwifi: mvm: don't WARN when statistics are handled late Eric Dumazet (3): pkt_sched: move the sanity test in qdisc_list_add() pkt_sched: fq: do not hold qdisc lock while allocating memory tcp: tcp_release_cb() should release socket ownership Eric W. Biederman (3): audit: Use struct net not pid_t to remember the network namespce to reply in audit: Send replies in the proper network namespace. audit: Update kdoc for audit_send_reply and audit_list_rules_send Erik Hugne (3): tipc: drop subscriber connection id invalidation tipc: fix memory leak during module removal tipc: don't log disabled tasklet handler errors Fernando Luis Vazquez Cao (1): sched/clock: Prevent tracing recursion in sched_clock_cpu() Florian Westphal (1): inet: frag: make sure forced eviction removes all frags Gavin Shan (1): net/mlx4: Support shutdown() interface Geert Uytterhoeven (2): cris: convert ffs from an object-like macro to a function-like macro packet: doc: Spelling s/than/that/ Giridhar Malavali (1): [SCSI] qla2xxx: Poll during initialization for ISP25xx and ISP83xx Giuseppe CAVALLARO (4): stmmac: disable at run-time the EEE if not supported stmmac: fix and better tune the default buffer sizes stmmac: fix chained mode stmmac: dwmac-sti: fix broken STiD127 compatibility Guillaume Nault (2): l2tp: fix manual sequencing (de)activation in L2TPv2 l2tp: fix userspace reception on plain L2TP sockets Haiyang Zhang (1): hyperv: Move state setting for link query Heiner Kallweit (1): ipv6: Avoid unnecessary temporary addresses being generated Heinz Mauelshagen (2): dm cache: fix truncation bug when copying a block to/from >2TB fast device dm cache: fix access beyond end of origin device Ingo Molnar (1): Merge tag 'perf-urgent-for-mingo' of git://git.kernel.org/.../acme/linux into perf/urgent Ivaylo Dimitrov (1): wl1251: use skb_trim to make skb shorter Jean-Francois Moine (2): ASoC: simple-card: Fix the reference count of device nodes ASoC: simple-card: Simplify code Jeff Layton (2): cifs: sanity check length of data to send before sending cifs: mask off top byte in get_rfc1002_length() Jiri Olsa (1): perf symbols: Fix crash in elf_section_by_name Johannes Berg (2): cfg80211: remove racy beacon_interval assignment mac80211: clear sequence/fragment number in QoS-null frames Johannes Weiner (1): mm: fix GFP_THISNODE callers and clarify John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Juri Lelli (1): sched/deadline: Deny unprivileged users to set/change SCHED_DEADLINE policy Jyri Sarha (1): ASoC: tlv320aic31xx: Turn power off only once. Kuninori Morimoto (4): ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: fsi: fixup compile error for simple-card ASoC: rsnd: add DeviceTree support Lars-Peter Clausen (8): ASoC: omap-abe-twl6040: Register machine level DMIC DAPM routes with the card ASoC: davinci-evm: Register machine level DAPM elements with the card ASoC: max98090: Remove unused control_data field ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call ASoC: codecs: Replace instances of rtd->codec with dai->codec ASoC: Add helper function to cast component back to CODEC ASoC: mfld_machine: Convert to table based DAPM and control setup ASoC: mfld_machine: Fix compile error Laura Abbott (1): mm/compaction: break out of loop on !PageBuddy in isolate_freepages_block Linus Lüssing (3): bridge: multicast: add sanity check for query source addresses bridge: multicast: add sanity check for general query destination bridge: multicast: enable snooping on general queries only Linus Torvalds (19): vfs: atomic f_pos accesses as per POSIX Merge branch 'for-3.14-fixes' of git://git.kernel.org/.../tj/libata Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'akpm' (patches from Andrew Morton) Merge branch 'for-linus' of git://git.kernel.org/.../ebiederm/user-namespace Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'sound-3.14-rc7' of git://git.kernel.org/.../tiwai/sound Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'pci-v3.14-fixes-2' of git://git.kernel.org/.../helgaas/pci Merge git://git.kernel.org/.../davem/net Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'dm-3.14-fixes-4' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'pm+acpi-3.14-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.14-rc7 Lukasz Dorau (1): [SCSI] isci: correct erroneous for_each_isci_host macro Marc Zyngier (1): ARM: KVM: fix non-VGIC compilation Mark Brown (13): ASoC: io: Remove support for ASoC cache in conjunction with regmap ASoC: pcm: Drop incorrect double/extra frees Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/codecs' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42xx8' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/io' and 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next Markos Chandras (1): tools/net/Makefile: Define PACKAGE to fix build problems Matthew Leach (1): net: socket: error on a negative msg_namelen Michael Chan (1): bnx2: Fix shutdown sequence Michael Kerrisk (1): ipc: Fix 2 bugs in msgrcv() MSG_COPY implementation Michael Opdenacker (1): MAINTAINERS: blackfin: add git repository Michael S. Tsirkin (5): skbuff: skb_segment: s/frag/nskb_frag/ skbuff: skb_segment: s/skb_frag/frag/ skbuff: skb_segment: s/skb/head_skb/ skbuff: skb_segment: s/fskb/list_skb/ skbuff: skb_segment: orphan frags before copying Mike Christie (1): [SCSI] be2iscsi: fix bad if expression Neil Horman (1): vmxnet3: fix netpoll race condition Nenghua Cao (1): asoc: soc-core: fix coccinelle warnings Nicolin Chen (2): ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt ASoC: cs42xx8: Add codec driver support for CS42448/CS42888 Nikolay Aleksandrov (1): net: fix for a race condition in the inet frag code Or Gerlitz (4): net/mlx4_core: Fix wrong dump of the vxlan offloads device capability net/mlx4_en: Handle vxlan steering rules for mac address changes net/mlx4_core: Load the IB driver when the device supports IBoE net/mlx4_en: Deregister multicast vxlan steering rules when going down Oren Givon (1): iwlwifi: fix and add 7265 series HW IDs Pavel Shilovsky (1): CIFS: Fix wrong pos argument of cifs_find_lock_conflict Peter Boström (1): vlan: Set correct source MAC address with TX VLAN offload enabled Peter Ujfalusi (2): ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot Peter Zijlstra (1): stop_machine: Fix^2 race between stop_two_cpus() and stop_cpus() Radim Krčmář (1): KVM: SVM: fix cr8 intercept window Rafael J. Wysocki (4): cpufreq: Skip current frequency initialization for ->setpolicy drivers ACPI / init: Invoke early ACPI initialization later ACPI / sleep: Add extra checks for HW Reduced ACPI mode sleep states Merge branches 'pnp', 'acpi-init', 'acpi-sleep' and 'pm-cpufreq' Richard Weinberger (1): i2c: Remove usage of orphaned symbol OF_I2C Rob Clark (1): drm/ttm: don't oops if no invalidate_caches() Sabrina Dubroca (1): ipv6: don't set DST_NOCOUNT for remotely added routes Sasha Levin (1): bonding: correctly handle out of range parameters for lp_interval Sergei Antonov (1): hfsplus: add HFSX subfolder count support Soren Brinkmann (2): net: macb: Check DMA mappings for error net: macb: DMA-unmap full rx-buffer Stefan Wahren (1): eth: fec: Fix lost promiscuous mode after reconnecting cable Suresh Siddha (1): x86, fpu: Check tsk_used_math() in kernel_fpu_end() for eager FPU Tejun Heo (1): libata: use wider match for blacklisting Crucial M500 Thomas Hellstrom (2): drm/ttm: Work around performance regression with VM_PFNMAP drm/vmwgfx: Fix a surface reference corner-case in legacy emulation mode Thomas Petazzoni (1): ASoC: add S/PDIF support to Armada 370 DB ASoC driver Tobias Klauser (1): MAINTAINERS: Add tools/net to NETWORKING [GENERAL] Wei Liu (1): xen-netback: use skb_is_gso in xenvif_start_xmit Ying Xue (3): tipc: allow connection shutdown callback to be invoked in advance tipc: fix connection refcount leak tipc: avoid to unnecessary process switch under non-block mode Zhang Rui (1): PNP / ACPI: proper handling of ACPI IO/Memory resource parsing failures dingtianhong (1): bonding: set correct vlan id for alb xmit path hayeswang (2): r8152: disable the ECM mode r8169: fix the incorrect tx descriptor version .gitignore | 3 + Documentation/00-INDEX | 24 +- Documentation/PCI/MSI-HOWTO.txt | 119 +- Documentation/RCU/00-INDEX | 2 + Documentation/arm/00-INDEX | 14 + Documentation/blackfin/00-INDEX | 6 +- Documentation/block/00-INDEX | 2 + 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arch/arm/mach-pxa/include/mach/idp.h | 1 + arch/arm/mach-pxa/include/mach/palmld.h | 2 + arch/arm/mach-pxa/include/mach/palmt5.h | 2 + arch/arm/mach-pxa/include/mach/palmtc.h | 2 + arch/arm/mach-pxa/include/mach/palmtx.h | 2 + arch/arm/mach-pxa/include/mach/pcm027.h | 2 + arch/arm/mach-pxa/include/mach/pcm990_baseboard.h | 1 + arch/arm/mach-pxa/include/mach/poodle.h | 2 + arch/arm/mach-pxa/include/mach/spitz.h | 2 +- arch/arm/mach-pxa/include/mach/tosa.h | 2 + arch/arm/mach-pxa/include/mach/trizeps4.h | 2 + arch/arm/mach-pxa/mioa701.c | 9 + arch/arm/mach-sa1100/include/mach/collie.h | 2 + arch/arm/mach-shmobile/Kconfig | 2 +- arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +- arch/arm/mach-shmobile/board-kzm9g.c | 4 +- arch/arm/mach-shmobile/board-mackerel.c | 6 +- arch/arm/mach-tegra/pm.c | 1 + arch/arm/mach-tegra/tegra.c | 10 + arch/arm/mach-zynq/common.c | 14 + arch/arm/mm/dma-mapping.c | 2 +- arch/arm/mm/dump.c | 3 + arch/arm/mm/mm.h | 1 + arch/arm/mm/mmu.c | 7 +- arch/arm/mm/proc-v6.S | 3 +- arch/arm/mm/proc-v7.S | 2 +- arch/arm64/Kconfig | 1 + arch/arm64/configs/defconfig | 18 +- arch/arm64/include/asm/atomic.h | 53 +- arch/arm64/include/asm/barrier.h | 2 +- arch/arm64/include/asm/cacheflush.h | 1 + arch/arm64/include/asm/cmpxchg.h | 17 +- arch/arm64/include/asm/esr.h | 2 +- arch/arm64/include/asm/futex.h | 10 +- arch/arm64/include/asm/kvm_arm.h | 2 +- arch/arm64/include/asm/percpu.h | 8 + arch/arm64/include/asm/pgtable.h | 10 +- arch/arm64/include/asm/spinlock.h | 10 +- arch/arm64/include/asm/unistd32.h | 5 +- arch/arm64/include/uapi/asm/kvm.h | 9 + arch/arm64/kernel/kuser32.S | 6 +- arch/arm64/kernel/stacktrace.c | 6 +- arch/arm64/kernel/vdso.c | 4 +- arch/arm64/kernel/vdso/Makefile | 2 +- arch/arm64/kernel/vdso/gettimeofday.S | 7 +- arch/arm64/kvm/hyp.S | 27 +- arch/arm64/lib/bitops.S | 3 +- arch/arm64/mm/dma-mapping.c | 1 + arch/arm64/mm/mmu.c | 12 +- arch/arm64/mm/pgd.c | 11 +- arch/avr32/Makefile | 2 +- arch/avr32/boards/mimc200/fram.c | 1 + arch/avr32/include/asm/Kbuild | 1 + arch/avr32/include/asm/io.h | 2 + arch/c6x/include/asm/cache.h | 1 + arch/cris/include/asm/bitops.h | 2 +- arch/ia64/include/asm/unistd.h | 2 +- arch/ia64/include/uapi/asm/unistd.h | 2 + arch/ia64/kernel/entry.S | 2 + arch/ia64/kernel/uncached.c | 2 +- arch/m68k/include/asm/Kbuild | 6 +- arch/m68k/include/asm/barrier.h | 8 - arch/m68k/include/asm/unistd.h | 2 +- arch/m68k/include/uapi/asm/unistd.h | 2 + arch/m68k/kernel/syscalltable.S | 2 + arch/microblaze/include/asm/delay.h | 2 + arch/microblaze/include/asm/io.h | 6 + arch/microblaze/kernel/head.S | 2 +- arch/mips/alchemy/devboards/db1000.c | 7 +- arch/mips/include/asm/fpu.h | 2 + arch/mips/include/uapi/asm/unistd.h | 18 +- arch/mips/kernel/scall32-o32.S | 2 + arch/mips/kernel/scall64-64.S | 2 + arch/mips/kernel/scall64-n32.S | 2 + arch/mips/kernel/scall64-o32.S | 2 + arch/parisc/hpux/fs.c | 15 +- arch/powerpc/include/asm/compat.h | 5 +- arch/powerpc/include/asm/dma-mapping.h | 1 + arch/powerpc/include/asm/eeh.h | 21 +- arch/powerpc/include/asm/hugetlb.h | 2 +- arch/powerpc/include/asm/iommu.h | 1 + arch/powerpc/include/asm/opal.h | 4 +- arch/powerpc/include/asm/pgtable-ppc64.h | 26 +- arch/powerpc/include/asm/pgtable.h | 22 + arch/powerpc/include/asm/ptrace.h | 16 +- arch/powerpc/include/asm/sections.h | 12 + arch/powerpc/include/asm/vdso.h | 6 +- arch/powerpc/kernel/crash_dump.c | 8 +- arch/powerpc/kernel/dma.c | 10 +- arch/powerpc/kernel/eeh.c | 32 +- arch/powerpc/kernel/eeh_driver.c | 8 +- arch/powerpc/kernel/ftrace.c | 1 + arch/powerpc/kernel/iommu.c | 12 + arch/powerpc/kernel/irq.c | 5 + arch/powerpc/kernel/machine_kexec.c | 14 +- arch/powerpc/kernel/machine_kexec_64.c | 6 +- arch/powerpc/kernel/misc_32.S | 5 +- arch/powerpc/kernel/process.c | 9 + arch/powerpc/kernel/reloc_64.S | 5 +- arch/powerpc/kernel/setup_32.c | 5 + arch/powerpc/kernel/signal_64.c | 4 +- arch/powerpc/kernel/vdso32/vdso32_wrapper.S | 2 +- arch/powerpc/kernel/vdso64/vdso64_wrapper.S | 2 +- arch/powerpc/mm/hash_utils_64.c | 14 + arch/powerpc/mm/pgtable_64.c | 12 +- arch/powerpc/mm/subpage-prot.c | 2 +- arch/powerpc/perf/core-book3s.c | 5 +- arch/powerpc/perf/power8-pmu.c | 144 ++ arch/powerpc/platforms/cell/ras.c | 3 +- arch/powerpc/platforms/powernv/eeh-ioda.c | 128 +- arch/powerpc/platforms/powernv/eeh-powernv.c | 2 +- arch/powerpc/platforms/powernv/opal-xscom.c | 21 +- arch/powerpc/platforms/powernv/pci-ioda.c | 84 + arch/powerpc/platforms/powernv/pci.c | 230 +-- arch/powerpc/platforms/powernv/pci.h | 6 +- arch/powerpc/platforms/powernv/powernv.h | 8 + arch/powerpc/platforms/powernv/setup.c | 9 + arch/powerpc/platforms/pseries/Kconfig | 1 + arch/powerpc/platforms/pseries/eeh_pseries.c | 2 +- arch/powerpc/platforms/pseries/hotplug-cpu.c | 22 +- arch/powerpc/platforms/pseries/pci.c | 22 +- arch/powerpc/platforms/pseries/setup.c | 3 +- arch/powerpc/sysdev/mpic.c | 38 +- arch/powerpc/xmon/xmon.c | 24 +- arch/s390/appldata/appldata_base.c | 1 + arch/s390/crypto/aes_s390.c | 65 +- arch/s390/crypto/des_s390.c | 95 +- arch/s390/kernel/compat_wrapper.S | 2 +- arch/s390/kernel/head64.S | 7 +- arch/s390/mm/page-states.c | 10 + arch/s390/pci/pci_dma.c | 8 +- arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- arch/sh/include/cpu-sh2/cpu/cache.h | 2 +- arch/sh/include/cpu-sh2a/cpu/cache.h | 4 +- arch/sh/include/cpu-sh3/cpu/cache.h | 2 +- arch/sh/include/cpu-sh4/cpu/cache.h | 2 +- arch/sh/kernel/cpu/init.c | 4 +- arch/sh/mm/cache-debugfs.c | 2 +- arch/sh/mm/cache-sh2.c | 4 +- arch/sh/mm/cache-sh2a.c | 6 +- arch/sh/mm/cache-sh4.c | 4 +- arch/sh/mm/cache-shx3.c | 4 +- arch/sh/mm/cache.c | 4 +- arch/sparc/Kconfig | 2 +- arch/sparc/mm/srmmu.c | 2 + arch/x86/Kconfig | 7 +- arch/x86/Kconfig.cpu | 4 - arch/x86/Kconfig.debug | 1 + arch/x86/boot/compressed/aslr.c | 9 +- arch/x86/include/asm/amd_nb.h | 2 +- arch/x86/include/asm/barrier.h | 8 +- arch/x86/include/asm/efi.h | 3 + arch/x86/include/asm/io.h | 2 +- arch/x86/include/asm/pgtable.h | 14 +- arch/x86/include/asm/spinlock.h | 5 +- arch/x86/include/asm/tlbflush.h | 6 +- arch/x86/include/asm/tsc.h | 2 +- arch/x86/include/asm/xen/page.h | 5 +- arch/x86/kernel/amd_nb.c | 2 +- arch/x86/kernel/cpu/amd.c | 5 +- arch/x86/kernel/cpu/centaur.c | 272 --- arch/x86/kernel/cpu/common.c | 7 +- arch/x86/kernel/cpu/intel.c | 10 +- arch/x86/kernel/cpu/microcode/amd_early.c | 43 +- arch/x86/kernel/cpu/mtrr/generic.c | 4 +- arch/x86/kernel/cpu/perf_event.c | 11 +- arch/x86/kernel/cpu/perf_event.h | 1 + arch/x86/kernel/cpu/perf_event_intel.c | 11 +- arch/x86/kernel/cpu/perf_event_intel_uncore.c | 13 +- arch/x86/kernel/cpu/perf_event_p6.c | 48 +- arch/x86/kernel/ftrace.c | 83 +- arch/x86/kernel/head_32.S | 7 +- arch/x86/kernel/head_64.S | 6 +- arch/x86/kernel/i387.c | 15 +- arch/x86/kernel/irq.c | 9 +- arch/x86/kernel/machine_kexec_64.c | 2 + arch/x86/kernel/pci-dma.c | 4 +- arch/x86/kernel/quirks.c | 39 +- arch/x86/kernel/setup.c | 10 +- arch/x86/kernel/tsc.c | 11 +- arch/x86/kernel/tsc_msr.c | 30 +- arch/x86/kvm/mmu.c | 1 + arch/x86/kvm/svm.c | 6 +- arch/x86/kvm/vmx.c | 2 +- arch/x86/kvm/x86.c | 2 +- arch/x86/mm/fault.c | 61 +- arch/x86/mm/numa.c | 21 +- arch/x86/mm/numa_32.c | 2 + arch/x86/mm/srat.c | 16 +- arch/x86/mm/tlb.c | 52 +- arch/x86/net/bpf_jit.S | 2 +- arch/x86/platform/efi/efi-bgrt.c | 12 +- arch/x86/platform/efi/efi.c | 25 +- arch/x86/platform/efi/efi_32.c | 6 + arch/x86/platform/efi/efi_64.c | 9 + arch/x86/um/asm/barrier.h | 4 - arch/x86/xen/enlighten.c | 12 + arch/x86/xen/mmu.c | 4 +- arch/x86/xen/p2m.c | 17 +- arch/xtensa/Kconfig | 3 +- arch/xtensa/boot/dts/xtfpga.dtsi | 12 +- arch/xtensa/include/asm/io.h | 2 +- arch/xtensa/include/asm/traps.h | 44 +- arch/xtensa/include/asm/vectors.h | 2 +- arch/xtensa/include/uapi/asm/unistd.h | 7 +- arch/xtensa/kernel/entry.S | 449 ++--- arch/xtensa/kernel/setup.c | 2 + arch/xtensa/kernel/time.c | 1 + arch/xtensa/kernel/vectors.S | 2 +- arch/xtensa/kernel/xtensa_ksyms.c | 2 - arch/xtensa/mm/init.c | 13 +- arch/xtensa/mm/mmu.c | 2 +- arch/xtensa/platforms/xtfpga/setup.c | 7 +- arch/xtensa/variants/fsf/include/variant/tie.h | 9 +- block/blk-core.c | 20 +- block/blk-exec.c | 2 +- block/blk-flush.c | 105 +- block/blk-lib.c | 8 + block/blk-merge.c | 91 +- block/blk-mq-cpu.c | 14 +- block/blk-mq-tag.c | 2 +- block/blk-mq.c | 237 ++- block/blk-mq.h | 5 +- block/blk-sysfs.c | 2 + block/blk-timeout.c | 2 +- block/blk.h | 2 +- drivers/acpi/ac.c | 2 + drivers/acpi/battery.c | 4 +- drivers/acpi/blacklist.c | 58 - drivers/acpi/button.c | 2 + drivers/acpi/container.c | 5 +- drivers/acpi/dock.c | 13 +- drivers/acpi/ec.c | 64 + drivers/acpi/fan.c | 3 + drivers/acpi/pci_irq.c | 1 + drivers/acpi/proc.c | 2 +- drivers/acpi/processor_throttling.c | 69 +- drivers/acpi/resource.c | 10 + drivers/acpi/sbs.c | 4 +- drivers/acpi/scan.c | 6 +- drivers/acpi/sleep.c | 32 +- drivers/acpi/thermal.c | 2 + drivers/acpi/utils.c | 4 - drivers/acpi/video.c | 147 +- drivers/acpi/video_detect.c | 8 - drivers/ata/Kconfig | 1 + drivers/ata/ahci.c | 18 +- drivers/ata/libata-core.c | 3 +- drivers/ata/libata-pmp.c | 7 +- drivers/ata/pata_imx.c | 8 +- drivers/ata/sata_mv.c | 16 +- drivers/ata/sata_sil.c | 1 + drivers/base/component.c | 8 + drivers/base/dma-buf.c | 25 +- drivers/base/firmware_class.c | 1 + drivers/base/regmap/regmap.c | 12 + drivers/block/aoe/aoecmd.c | 4 +- drivers/block/mtip32xx/mtip32xx.h | 2 +- drivers/block/null_blk.c | 97 +- drivers/block/nvme-core.c | 610 +++++-- drivers/block/nvme-scsi.c | 147 ++ drivers/block/virtio_blk.c | 7 +- drivers/block/xen-blkback/blkback.c | 81 +- drivers/block/xen-blkback/common.h | 5 +- drivers/block/xen-blkback/xenbus.c | 14 + drivers/block/xen-blkfront.c | 11 +- drivers/block/zram/zram_drv.c | 2 + drivers/char/Kconfig | 1 + drivers/char/raw.c | 2 +- drivers/char/virtio_console.c | 9 +- drivers/clk/at91/clk-master.c | 2 +- drivers/clk/clk-nomadik.c | 3 + drivers/clk/clk.c | 13 +- drivers/clk/keystone/gate.c | 1 + drivers/clk/mvebu/armada-370.c | 21 +- drivers/clk/mvebu/armada-xp.c | 20 +- drivers/clk/mvebu/dove.c | 19 +- drivers/clk/mvebu/kirkwood.c | 34 +- drivers/clk/shmobile/clk-rcar-gen2.c | 48 +- drivers/clk/tegra/clk-divider.c | 2 +- drivers/clk/tegra/clk-id.h | 4 + drivers/clk/tegra/clk-tegra-periph.c | 10 +- drivers/clk/tegra/clk-tegra-super-gen4.c | 2 +- drivers/clk/tegra/clk-tegra114.c | 8 +- drivers/clk/tegra/clk-tegra124.c | 48 +- drivers/clk/tegra/clk-tegra20.c | 2 + drivers/clocksource/bcm_kona_timer.c | 54 +- drivers/cpufreq/cpufreq.c | 58 +- drivers/cpufreq/intel_pstate.c | 61 +- drivers/cpufreq/powernow-k8.c | 10 +- drivers/crypto/nx/nx-842.c | 29 +- drivers/dma/Kconfig | 1 + drivers/dma/imx-sdma.c | 1 + drivers/dma/ioat/dma.c | 52 +- drivers/dma/ioat/dma.h | 1 + drivers/dma/ioat/dma_v2.c | 11 +- drivers/dma/ioat/dma_v3.c | 3 + drivers/dma/mv_xor.c | 24 +- drivers/dma/ste_dma40.c | 4 +- drivers/edac/edac_mc.c | 13 +- drivers/edac/edac_mc_sysfs.c | 10 +- drivers/edac/edac_module.h | 2 +- drivers/edac/i7300_edac.c | 38 +- drivers/edac/i7core_edac.c | 9 +- drivers/extcon/extcon-arizona.c | 12 - drivers/firewire/core-device.c | 22 +- drivers/firewire/net.c | 6 +- drivers/firewire/ohci.c | 15 +- drivers/firewire/sbp2.c | 17 +- drivers/fmc/fmc-write-eeprom.c | 2 +- drivers/gpio/Kconfig | 1 + drivers/gpio/gpio-bcm-kona.c | 4 +- drivers/gpio/gpio-clps711x.c | 1 + drivers/gpio/gpio-intel-mid.c | 4 +- drivers/gpio/gpio-xtensa.c | 16 + drivers/gpu/drm/armada/armada_drv.c | 10 +- drivers/gpu/drm/ast/ast_fb.c | 2 +- drivers/gpu/drm/bochs/Kconfig | 1 + drivers/gpu/drm/cirrus/cirrus_fbdev.c | 2 +- drivers/gpu/drm/drm_ioctl.c | 12 + drivers/gpu/drm/exynos/Kconfig | 4 +- drivers/gpu/drm/exynos/exynos_drm_drv.c | 14 +- drivers/gpu/drm/exynos/exynos_drm_g2d.c | 2 +- drivers/gpu/drm/exynos/exynos_drm_ipp.c | 3 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 66 +- drivers/gpu/drm/i2c/tda998x_drv.c | 19 +- drivers/gpu/drm/i915/i915_drv.c | 23 +- drivers/gpu/drm/i915/i915_drv.h | 8 + drivers/gpu/drm/i915/i915_gem_stolen.c | 19 +- drivers/gpu/drm/i915/i915_gpu_error.c | 5 +- drivers/gpu/drm/i915/i915_irq.c | 3 +- drivers/gpu/drm/i915/intel_display.c | 22 +- drivers/gpu/drm/i915/intel_dp.c | 31 +- drivers/gpu/drm/i915/intel_hdmi.c | 6 +- drivers/gpu/drm/i915/intel_i2c.c | 7 - drivers/gpu/drm/i915/intel_opregion.c | 9 +- drivers/gpu/drm/i915/intel_panel.c | 4 +- drivers/gpu/drm/i915/intel_pm.c | 6 +- drivers/gpu/drm/i915/intel_ringbuffer.c | 21 + drivers/gpu/drm/i915/intel_ringbuffer.h | 1 + drivers/gpu/drm/mgag200/mgag200_fb.c | 2 +- drivers/gpu/drm/mgag200/mgag200_mode.c | 4 +- drivers/gpu/drm/msm/mdp/mdp4/mdp4_crtc.c | 179 +- drivers/gpu/drm/msm/mdp/mdp4/mdp4_plane.c | 4 +- drivers/gpu/drm/msm/mdp/mdp5/mdp5_crtc.c | 7 +- drivers/gpu/drm/msm/msm_gem.c | 2 +- drivers/gpu/drm/msm/msm_gem_submit.c | 9 +- drivers/gpu/drm/msm/msm_gpu.c | 3 - drivers/gpu/drm/nouveau/Makefile | 1 + drivers/gpu/drm/nouveau/core/engine/device/nv40.c | 10 +- drivers/gpu/drm/nouveau/core/engine/disp/nv50.c | 2 +- drivers/gpu/drm/nouveau/core/engine/fifo/nve0.c | 2 +- drivers/gpu/drm/nouveau/core/engine/graph/nv50.c | 2 +- drivers/gpu/drm/nouveau/core/include/subdev/mc.h | 1 + drivers/gpu/drm/nouveau/core/subdev/bios/base.c | 4 + drivers/gpu/drm/nouveau/core/subdev/fb/nv1a.c | 2 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv04.h | 1 + drivers/gpu/drm/nouveau/core/subdev/mc/nv44.c | 2 +- drivers/gpu/drm/nouveau/core/subdev/mc/nv4c.c | 45 + drivers/gpu/drm/nouveau/nouveau_acpi.c | 26 +- drivers/gpu/drm/nouveau/nouveau_bo.c | 2 +- drivers/gpu/drm/nouveau/nouveau_drm.c | 3 + drivers/gpu/drm/nouveau/nouveau_vga.c | 4 +- drivers/gpu/drm/radeon/atombios_crtc.c | 31 +- drivers/gpu/drm/radeon/atombios_encoders.c | 9 +- drivers/gpu/drm/radeon/btc_dpm.c | 32 + drivers/gpu/drm/radeon/btcd.h | 4 + drivers/gpu/drm/radeon/cik.c | 10 +- drivers/gpu/drm/radeon/cik_sdma.c | 14 +- drivers/gpu/drm/radeon/dce6_afmt.c | 15 +- drivers/gpu/drm/radeon/evergreen.c | 7 +- drivers/gpu/drm/radeon/evergreen_hdmi.c | 26 +- drivers/gpu/drm/radeon/evergreen_smc.h | 2 +- drivers/gpu/drm/radeon/kv_dpm.c | 2 +- drivers/gpu/drm/radeon/ni.c | 3 +- drivers/gpu/drm/radeon/ni_dpm.c | 10 +- drivers/gpu/drm/radeon/r100.c | 2 - drivers/gpu/drm/radeon/r300.c | 2 - drivers/gpu/drm/radeon/r420.c | 2 - drivers/gpu/drm/radeon/r520.c | 2 - drivers/gpu/drm/radeon/r600.c | 7 +- drivers/gpu/drm/radeon/r600_audio.c | 14 +- drivers/gpu/drm/radeon/r600_cs.c | 18 +- drivers/gpu/drm/radeon/r600_hdmi.c | 15 +- drivers/gpu/drm/radeon/radeon.h | 10 +- drivers/gpu/drm/radeon/radeon_asic.c | 2 +- drivers/gpu/drm/radeon/radeon_asic.h | 2 + drivers/gpu/drm/radeon/radeon_atpx_handler.c | 3 +- drivers/gpu/drm/radeon/radeon_device.c | 5 +- drivers/gpu/drm/radeon/radeon_display.c | 2 + drivers/gpu/drm/radeon/radeon_drv.c | 3 +- drivers/gpu/drm/radeon/radeon_kms.c | 16 +- drivers/gpu/drm/radeon/radeon_ring.c | 2 +- drivers/gpu/drm/radeon/radeon_semaphore.c | 19 +- drivers/gpu/drm/radeon/radeon_ttm.c | 5 +- drivers/gpu/drm/radeon/radeon_uvd.c | 2 + drivers/gpu/drm/radeon/reg_srcs/r600 | 1 + drivers/gpu/drm/radeon/rs400.c | 2 - drivers/gpu/drm/radeon/rs600.c | 2 - drivers/gpu/drm/radeon/rs690.c | 2 - drivers/gpu/drm/radeon/rv515.c | 2 - drivers/gpu/drm/radeon/rv770.c | 5 +- drivers/gpu/drm/radeon/rv770_dpm.c | 14 +- drivers/gpu/drm/radeon/si.c | 7 +- drivers/gpu/drm/radeon/si_dpm.c | 5 +- drivers/gpu/drm/radeon/sumo_dpm.c | 2 +- drivers/gpu/drm/radeon/trinity_dpm.c | 3 +- drivers/gpu/drm/radeon/uvd_v2_2.c | 1 - drivers/gpu/drm/tegra/drm.c | 2 +- drivers/gpu/drm/tegra/rgb.c | 11 + drivers/gpu/drm/ttm/ttm_agp_backend.c | 1 + drivers/gpu/drm/ttm/ttm_bo.c | 8 +- drivers/gpu/drm/ttm/ttm_bo_vm.c | 12 +- drivers/gpu/drm/ttm/ttm_object.c | 2 +- drivers/gpu/drm/ttm/ttm_tt.c | 3 + drivers/gpu/drm/vmwgfx/svga3d_reg.h | 153 +- drivers/gpu/drm/vmwgfx/svga3d_surfacedefs.h | 11 +- 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net/sctp/ulpevent.c | 8 +- net/socket.c | 17 +- net/sunrpc/auth_gss/auth_gss.c | 19 +- net/sunrpc/backchannel_rqst.c | 6 +- net/sunrpc/svc_xprt.c | 6 +- net/sunrpc/xprtsock.c | 6 +- net/tipc/bearer.c | 7 +- net/tipc/config.c | 11 +- net/tipc/core.c | 109 +- net/tipc/core.h | 2 +- net/tipc/handler.c | 1 - net/tipc/link.c | 7 + net/tipc/name_table.c | 40 +- net/tipc/netlink.c | 8 - net/tipc/ref.c | 3 - net/tipc/server.c | 19 +- net/tipc/server.h | 2 - net/tipc/socket.c | 12 +- net/tipc/subscr.c | 19 +- net/unix/af_unix.c | 3 +- net/wireless/core.c | 19 +- net/wireless/core.h | 4 +- net/wireless/nl80211.c | 32 +- net/wireless/nl80211.h | 8 +- net/wireless/reg.c | 12 +- net/wireless/scan.c | 40 +- net/wireless/sme.c | 2 +- net/xfrm/xfrm_policy.c | 2 +- net/xfrm/xfrm_state.c | 23 +- net/xfrm/xfrm_user.c | 5 - scripts/Makefile.lib | 1 + scripts/checkpatch.pl | 4 +- scripts/gen_initramfs_list.sh | 2 +- scripts/get_maintainer.pl | 2 +- scripts/mod/file2alias.c | 4 +- scripts/mod/modpost.c | 13 + security/Kconfig | 2 +- security/keys/keyring.c | 6 +- security/selinux/nlmsgtab.c | 2 + security/selinux/ss/policydb.c | 8 +- security/selinux/ss/services.c | 4 + sound/core/pcm_lib.c | 1 + sound/pci/hda/hda_codec.c | 34 +- sound/pci/hda/hda_generic.c | 8 +- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/patch_analog.c | 31 + sound/pci/hda/patch_ca0132.c | 68 +- sound/pci/hda/patch_conexant.c | 3 +- sound/pci/hda/patch_realtek.c | 92 +- sound/pci/hda/patch_sigmatel.c | 60 +- sound/pci/hda/thinkpad_helper.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 20 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 120 +- sound/soc/codecs/Kconfig | 191 ++- sound/soc/codecs/Makefile | 33 + sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 154 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/adau1373.c | 39 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 159 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 24 +- sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 250 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 120 +- sound/soc/codecs/alc5632.c | 48 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 99 +- sound/soc/codecs/cs42l52.c | 109 +- sound/soc/codecs/cs42l73.c | 72 +- sound/soc/codecs/cs42xx8-i2c.c | 64 + sound/soc/codecs/cs42xx8.c | 602 +++++++ sound/soc/codecs/cs42xx8.h | 238 +++ sound/soc/codecs/da7210.c | 28 +- sound/soc/codecs/da7213.c | 159 +- sound/soc/codecs/da732x.c | 208 ++- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 111 +- sound/soc/codecs/isabelle.c | 71 +- sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 47 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 47 +- sound/soc/codecs/max98090.c | 202 +-- sound/soc/codecs/max98090.h | 1 - sound/soc/codecs/max98095.c | 60 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 30 +- sound/soc/codecs/ml26124.c | 22 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 84 +- sound/soc/codecs/rt5640.c | 87 +- sound/soc/codecs/sgtl5000.c | 18 +- sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sirf-audio-codec.c | 524 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 46 +- sound/soc/codecs/ssm2518.c | 24 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 180 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta32x.c | 90 +- sound/soc/codecs/sta529.c | 15 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 59 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 79 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 7 +- sound/soc/codecs/tlv320aic31xx.c | 19 +- sound/soc/codecs/tlv320aic32x4.c | 2 - sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 35 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 43 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 25 +- sound/soc/codecs/wm5100.c | 43 +- sound/soc/codecs/wm5102.c | 32 +- sound/soc/codecs/wm5110.c | 22 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 37 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 11 +- sound/soc/codecs/wm8580.c | 9 +- sound/soc/codecs/wm8711.c | 8 +- sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8737.c | 56 +- sound/soc/codecs/wm8741.c | 40 +- sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 12 +- sound/soc/codecs/wm8770.c | 10 +- sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 10 +- sound/soc/codecs/wm8900.c | 52 +- sound/soc/codecs/wm8903.c | 118 +- sound/soc/codecs/wm8904.c | 86 +- sound/soc/codecs/wm8940.c | 26 +- sound/soc/codecs/wm8955.c | 19 +- sound/soc/codecs/wm8958-dsp2.c | 10 +- sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 23 +- sound/soc/codecs/wm8962.c | 87 +- sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 10 +- sound/soc/codecs/wm8978.c | 38 +- sound/soc/codecs/wm8983.c | 51 +- sound/soc/codecs/wm8985.c | 46 +- sound/soc/codecs/wm8988.c | 70 +- sound/soc/codecs/wm8990.c | 49 +- sound/soc/codecs/wm8991.c | 52 +- sound/soc/codecs/wm8993.c | 74 +- sound/soc/codecs/wm8994.c | 186 +- sound/soc/codecs/wm8995.c | 50 +- sound/soc/codecs/wm8996.c | 87 +- sound/soc/codecs/wm8997.c | 29 +- sound/soc/codecs/wm9081.c | 34 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 81 +- sound/soc/davinci/davinci-mcasp.c | 294 ++-- sound/soc/davinci/edma-pcm.c | 57 + sound/soc/davinci/edma-pcm.h | 25 + sound/soc/fsl/Kconfig | 12 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 36 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/fsl_sai.c | 332 ++-- sound/soc/fsl/fsl_sai.h | 48 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/imx-sgtl5000.c | 10 +- sound/soc/fsl/imx-wm8962.c | 11 +- sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 295 ++-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 108 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 26 +- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 60 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 67 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/samsung/Kconfig | 8 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 229 ++- sound/soc/sh/rcar/core.c | 426 +++-- sound/soc/sh/rcar/gen.c | 113 +- sound/soc/sh/rcar/rsnd.h | 206 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 723 ++++++++ sound/soc/sh/rcar/ssi.c | 388 ++--- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 552 +++--- sound/soc/soc-dapm.c | 604 +++---- sound/soc/soc-io.c | 99 +- sound/soc/soc-jack.c | 5 +- sound/soc/soc-pcm.c | 109 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 8 +- sound/usb/Kconfig | 1 + sound/usb/mixer.c | 1 + sound/usb/mixer_maps.c | 9 + tools/lib/lockdep/Makefile | 6 +- tools/lib/lockdep/preload.c | 2 +- tools/lib/lockdep/run_tests.sh | 0 tools/lib/lockdep/uinclude/asm/hash.h | 6 + tools/lib/lockdep/uinclude/linux/rcu.h | 5 + tools/net/Makefile | 2 +- tools/perf/builtin-buildid-cache.c | 33 +- tools/perf/builtin-record.c | 10 +- tools/perf/builtin-report.c | 40 +- tools/perf/builtin-top.c | 6 +- tools/perf/builtin-trace.c | 32 +- tools/perf/config/Makefile | 2 +- tools/perf/config/feature-checks/Makefile | 2 +- tools/perf/design.txt | 1 - tools/perf/perf.h | 4 +- tools/perf/tests/vmlinux-kallsyms.c | 10 - tools/perf/util/annotate.c | 9 +- tools/perf/util/annotate.h | 2 + tools/perf/util/event.c | 36 +- tools/perf/util/event.h | 6 +- tools/perf/util/include/asm/hash.h | 6 + tools/perf/util/include/linux/bitops.h | 4 +- tools/perf/util/machine.c | 44 +- tools/perf/util/machine.h | 2 + tools/perf/util/map.c | 5 +- tools/perf/util/map.h | 1 + tools/perf/util/parse-events.c | 17 +- tools/perf/util/probe-event.c | 2 +- tools/perf/util/session.c | 6 + tools/perf/util/symbol-elf.c | 10 +- tools/perf/util/symbol.c | 67 +- tools/testing/selftests/ipc/msgque.c | 1 + virt/kvm/arm/vgic.c | 1 + virt/kvm/coalesced_mmio.c | 8 +- 1779 files changed, 37413 insertions(+), 15444 deletions(-) create mode 100644 Documentation/devicetree/bindings/interrupt-controller/lsi,zevio-intc.txt create mode 100644 Documentation/devicetree/bindings/net/opencores-ethoc.txt create mode 100644 Documentation/devicetree/bindings/net/sti-dwmac.txt rename Documentation/devicetree/bindings/pinctrl/{brcm,capri-pinctrl.txt => brcm,bcm11351-pinctrl.txt} (98%) create mode 100644 Documentation/devicetree/bindings/power/bq2415x.txt create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt delete mode 100644 Documentation/networking/3c505.txt create mode 100644 Documentation/spi/00-INDEX create mode 100644 Documentation/zh_CN/arm64/tagged-pointers.txt create mode 100644 arch/arm/boot/dts/at91-sama5d3_xplained.dts create mode 100644 arch/arm/boot/dts/omap3-overo-storm-tobi.dts rename arch/arm/boot/dts/{omap3-tobi.dts => omap3-overo-tobi-common.dtsi} (94%) create mode 100644 arch/arm/boot/dts/omap3-overo-tobi.dts delete mode 100644 arch/arm/boot/dts/testcases/tests.dtsi delete mode 100644 arch/m68k/include/asm/barrier.h create mode 100644 drivers/gpu/drm/nouveau/core/subdev/mc/nv4c.c create mode 100644 drivers/irqchip/irq-zevio.c create mode 100644 drivers/net/ethernet/stmicro/stmmac/dwmac-sti.c delete mode 100644 drivers/net/irda/ep7211-sir.c delete mode 100644 drivers/net/usb/r815x.c create mode 100644 drivers/net/usb/sr9800.c create mode 100644 drivers/net/usb/sr9800.h create mode 100644 drivers/of/testcase-data/testcases.dtsi rename {arch/arm/boot/dts/testcases => drivers/of/testcase-data}/tests-interrupts.dtsi (100%) create mode 100644 drivers/of/testcase-data/tests-match.dtsi rename {arch/arm/boot/dts/testcases => drivers/of/testcase-data}/tests-phandle.dtsi (100%) delete mode 100644 drivers/xen/xencomm.c create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/net/netfilter/nft_reject.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h rename include/{ => uapi}/xen/gntalloc.h (100%) rename include/{ => uapi}/xen/gntdev.h (100%) delete mode 100644 include/xen/interface/xencomm.h delete mode 100644 include/xen/xencomm.h create mode 100644 net/ipv4/netfilter/nft_reject_ipv4.c create mode 100644 net/ipv6/netfilter/nft_reject_ipv6.c create mode 100644 net/netfilter/nft_reject_inet.c create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) create mode 100644 sound/soc/kirkwood/armada-370-db.c delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c mode change 100644 => 100755 tools/lib/lockdep/run_tests.sh create mode 100644 tools/lib/lockdep/uinclude/asm/hash.h create mode 100644 tools/perf/util/include/asm/hash.h

2 1

Re: [alsa-devel] [PATCH 42/44] bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series
by Takashi Sakamoto 24 Mar '14

24 Mar '14

Hi Euan, Thanks for your testing. (Mar 24 2014 10:41), Euan de Kock wrote: > Thanks for the update. This bit me last night, and I just commented out > the check to keep working. > > I'll try and retest it tonight with this version. (On a Echo Audiofire > Pre8 with firmware < 5) > > Regards, > > Euan deKock. When reading this message, I wondered why this bug affects driving on Echo AudioFirePre8. As describing the title, this affects a part of BeBoB based devices. Did you test with M-Audio Firewire 1814 or M-Audio ProjectMix I/O? If not, it may be a bug for Fireworks driver. Then please send me output from: /proc/asound/cardX/firewire/*. (The 'cardX' should be for your device.) Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

Re: [alsa-devel] [PATCH 3.13 110/149] ASoC: pcm: free path list before exiting from error conditions
by Mark Brown 22 Mar '14

22 Mar '14

On Sat, Mar 22, 2014 at 03:53:08PM +0000, Ben Hutchings wrote: > On Thu, 2014-03-20 at 17:04 -0700, Greg Kroah-Hartman wrote: If you spot something on a stable patch that applies upstream (as opposed to being stable process or to do with the older kernel which are the most common things with these mails) please add both the maintainers and relevant lists. -stable mail only goes to signoffs so misses both lists and additional maintainers meaning things may not be going to the people who should see them (in this case it's Liam who mostly looks at DPCM for example). It might make sense for the -stable mails to do this in general, at least with the lists if not the maintainers. Ideally it's helpful to also send it in a separate thread since the volume of mail about -stable stuff is extremely high so things are easily missed but that's a bit more fun to do, I'm really not sure there's any general way to resolve that one sensibly. > > dpcm_path_get() allocates dynamic memory to hold path list. > > Corresponding dpcm_path_put() must be called to free the memory. > > dpcm_path_put() is not called under several error conditions. > > This leads to memory leak. > This is broken. dpcm_path_get() may return -ENOMEM and not initialise > the list at all. > If snd_soc_dapm_dai_get_connected_widgets() can fail (I don't think it > can) then dpcm_path_get() should be responsible for freeing the list > before returning. It can't fail without memory corruption or internal bugs and would only fail with a crash. > [...] > > --- a/sound/soc/soc-pcm.c > > +++ b/sound/soc/soc-pcm.c > [...] > > @@ -1979,6 +1981,7 @@ static int dpcm_fe_dai_open(struct snd_p > > fe->dpcm[stream].runtime = fe_substream->runtime; > > > > if (dpcm_path_get(fe, stream, &list) <= 0) { > > + dpcm_path_put(&list); > This is the one place where a memory leak seems to be possible, but the > < 0 and == 0 cases need to be distinguished. > Greg, please drop this until it is fixed properly upstream. It's actually not going to cause a leak there at all since we have an unconditional run through the rest of the function to a double free with references to the list that gets freed, AFAICT there wasn't a leak to start off with. I think what the function needs to do here is bomb out early on -ENOMEM and otherwise trundle on. Anyway, I'll revert this upstream. Thanks for noticing this.

1 0

[alsa-devel] [PATCH] ASoC: pcm: Drop incorrect double/extra frees
by Mark Brown 22 Mar '14

22 Mar '14

From: Mark Brown <broonie(a)linaro.org> The changes in "ASoC: pcm: free path list before exiting from error conditions" actually introduced both double frees (in case where the path list was allocated but empty) and frees of unallocated memory (in cases where the error being handled was -ENOMEM. Drop the commit for now. Fixes: e4ad1accb (ASoC: pcm: free path list before exiting from error conditions) Reported-by: Ben Hutchings <ben(a)decadent.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-pcm.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 28522bd03b8e..47e1ce771e65 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,7 +1989,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2019,7 +2018,6 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2084,7 +2082,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- 1.9.1

1 0

[alsa-devel] Edirol DA2496 and word clock sync
by Brian McNeil 22 Mar '14

22 Mar '14

Hello all; I'm new to the list, and sitting playing with a pair of Edirol DA2496 cards. These support Word Clock synchronisation, which can be done using S/PDIF, a 75 ohm coax cable on the breakout boxes, or with a internal jumper cable between the PCI cards. Unfortunately neither the driver, or Mudita24, were exposing access to the Word Clock. I've managed to patch support into both; doing so with the 3.12.13-rt21 kernel, and mudita24-1.0.3+svn13 from the Ubuntu repositories. This is, mostly, by treating the cards the same as the Delta1010LT. Forgive me being so rusty with C that I've not provided a patch; going to have to redo this and actually build proper diffs for it. I do have a related query, about Xruns - when using an .asoundrc file to merge the two cards into a single virtual card - is there any way to properly suppress backend Xruns that have no impact on the actual recorded audio? Brian McNeil -- Wikinewsie.org | http://wikinewsie.org | https://en.wikinews.org "Facts don't cease to be facts, but news ceases to be news." GPG Fingerprint: 7C3D FFD5 5ED5 B80F 1D18 A52B E84C 8928 6ABC A7AD

1 0

[alsa-devel] [PATCH] ASoC: mfld_machine: Fix compile error
by Lars-Peter Clausen 21 Mar '14

21 Mar '14

Fixes: 115f3f8 ("ASoC: mfld_machine: Convert to table based DAPM and control setup") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/mfld_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 4616942..031d787 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -254,7 +254,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { -- 1.8.0

2 1

[alsa-devel] [PATCH - ALSA-LIB 1/1] Remove 2.1 , 4.1 and 5.0 from EMU10K1.conf
by Raymond Yau 21 Mar '14

21 Mar '14

- Creative 4.1 Speaker only have green and black jacks - Those creative sound cards using 4.1 speakers does not support 6 channels

2 1

[alsa-devel] Ambiguous DAPM widget names and DPCM
by Daniel Mack 21 Mar '14

21 Mar '14

Hi Mark, Hi Liam, I spent some hours reverse-engineering the DPCM code, and I plan to prepare a patch to add some documentation once I've assembled a fully working setup. However, I have a general question regarding DAPM widget names. DPCM uses DAPM widgets and routes in order to determine the FE <-> BE connections at runtime. The problem with this is that widget names are very ambiguous, as the automatically created widgets for DAIs are named after their stream_name properties, which contain "Playback" or "Capture" for all components in my case. Hence, the code which walks the connections (snd_soc_dapm_dai_get_connected_widgets()) will most likely match the wrong one (which has empty sound/sink lists), which results in a failure like this: [ 16.606268] fe: ASoC: fe no valid playback route I hacked the names of all components so they are unique, but even then, the problem is that snd_soc_dapm_new_dai_widgets() is called twice for the CPU DAI, once from soc_probe_platform(), and later from soc_probe_link_dais(), and both calls would create DAPM widgets with identical names, resulting in the same problem as described above. I don't know how to handle this, but I believe that nobody was really hit by this issue yet, as DPCM doesn't seem to be used widely, at least not by any machine code in mainline. One idea would be to change the automatically generated names of DAI widgets in order to make them unique, or re-factor the lookup routines. Any opinion on that? Do I miss a general consideration here? With more hacks in place, the setup somehow succeeds, but when the stream is opened, we'll hit a BUG() in sound/core/pcm_memory.c, because the FE device's DMA type is undefined. But that's another topic I'll address later. Thanks, Daniel

5 9

[alsa-devel] [PATCH v5] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 21 Mar '14

21 Mar '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. As a result, standard regmap is difficult to be used on ALC286. We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. Now we have ,reg_write and .reg_read functions for ALC286's I/O. And we don't use cache due to item 3 above. Some dummy registers (address <= 0xff) are defined for dapm routing. Thhe dummy registers are cache only. Due to item 2 above, HD-A verb commands are put into the address part of regmap. When we issue HD-A verb write commands, the data part of regmap is zero. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Include <sound/hda_verbs.h> instead of hda_codec.h * Fix rt286_update_bits doesn't write the new value to codec issue. * Apply different settings for system clock = 19.2MHz and 24MHz. * Use .set_bclk_ratio instead of .set_dfs. * Change ACPI number from INT33CA to INT343A. * Other minor changes. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1379 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 159 ++++++ 5 files changed, 1563 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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 __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..c46e722 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -379,6 +380,9 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..ee2f368 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -204,6 +205,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..4621b62 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1379 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#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 <linux/spi/spi.h> +#include <linux/acpi.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 <sound/tlv.h> +#include <sound/jack.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct snd_soc_jack *jack; + struct i2c_client *i2c; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [RT286_AUDIO_FUNCTION_GROUP] = 0x0000, + [RT286_DAC_OUT1] = 0x7f7f, + [RT286_DAC_OUT2] = 0x7f7f, + [RT286_SPDIF] = 0x0000, + [RT286_ADC_IN1] = 0x4343, + [RT286_ADC_IN2] = 0x4343, + [RT286_MIC1] = 0x0000, + [RT286_MIXER_IN] = 0x000b, + [RT286_MIXER_OUT1] = 0x0002, + [RT286_MIXER_OUT2] = 0x0000, + [RT286_DMIC1] = 0x0000, + [RT286_DMIC2] = 0x0000, + [RT286_LINE1] = 0x0000, + [RT286_BEEP] = 0x0000, + [RT286_VENDOR_REGISTERS] = 0x0000, + [RT286_SPK_OUT] = 0x8080, + [RT286_HP_OUT] = 0x8080, + [RT286_MIXER_IN1] = 0x0000, + [RT286_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int old, new, verb; + int change, ret; + + verb = VERB_CMD((vid | 0x800), nid, data); + regmap_read(rt286->regmap, verb, &old); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + verb = VERB_CMD(vid, nid, new); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + verb = VERB_CMD(AC_VERB_SET_PROC_COEF, wid, data); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb, val; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + + verb = VERB_CMD(AC_VERB_GET_PROC_COEF, wid, index); + regmap_read(rt286->regmap, verb, &val); + return val; +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { RT286_DAC_OUT1, 0x7f7f }, + { RT286_DAC_OUT2, 0x7f7f }, + { RT286_SPDIF, 0x0000 }, + { RT286_ADC_IN1, 0x4343 }, + { RT286_ADC_IN2, 0x4343 }, + { RT286_MIC1, 0x0000 }, + { RT286_MIXER_IN, 0x000b }, + { RT286_MIXER_OUT1, 0x0002 }, + { RT286_MIXER_OUT2, 0x0000 }, + { RT286_SPK_OUT, 0x0000 }, + { RT286_HP_OUT, 0x0000 }, + { RT286_MIXER_IN1, 0x0005 }, + { RT286_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT286_DAC_OUT1: + case RT286_DAC_OUT2: + case RT286_ADC_IN1: + case RT286_ADC_IN2: + case RT286_MIC1: + case RT286_SPDIF: + case RT286_MIXER_IN: + case RT286_MIXER_OUT1: + case RT286_MIXER_OUT2: + case RT286_DMIC1: + case RT286_DMIC2: + case RT286_SPK_OUT: + case RT286_HP_OUT: + case RT286_MIXER_IN1: + case RT286_MIXER_IN2: + return true; + default: + return false; + } +} + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL2, 0x4, 0x4); + regmap_write(rt286->regmap, RT286_SET_MIC1_24, 0); + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + regmap_write(rt286->regmap, + RT286_SET_MIC1_20, 0); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x1001); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_IRQ_CTRL, 0x1, 0x1); + return 0; +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf, io = 0x8000; + + if (RT286_ADC_IN1 == mc->reg) + io = 0; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, (0x2000 | io)); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x7f; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, io); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = buf & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, io; + + if (RT286_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0x2000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x1000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x0); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + regmap_write(rt286->regmap, verb, 0); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0xa000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = !(buf & 0x80); + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x8000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = !(buf & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0xa000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x9000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", RT286_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", RT286_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", RT286_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", RT286_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", RT286_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", RT286_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", RT286_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SPK_OUT, + RT286_M_SPK_MUX_SFT, 1, 1); + +static const struct snd_kcontrol_new hpo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_HP_OUT, + RT286_M_HP_MUX_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, RT286_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_SPK_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SPK_EAPD_HIGH, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SPK_EAPD_LOW, 0); + regmap_write(rt286->regmap, RT286_DISABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SET_SPK_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_read(rt286->regmap, RT286_HP_OUT, &buf); + val = buf & 0x8080; + switch (val) { + case 0x0: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO, 0); + break; + case 0x8000: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_R, 0); + break; + case 0x0080: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_L, 0); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_MUTE_HPO, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_DMIC1_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_DMIC1, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_DMIC1, 0); + regmap_write(rt286->regmap, RT286_SET_DMIC1_D3, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_HPO_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_HPO, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_HPO, 0); + regmap_write(rt286->regmap, RT286_SET_HPO_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & RT286_ADC_SEL_MASK; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & 0x1; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", RT286_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, RT286_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, RT286_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH_E("HPO", SND_SOC_NOPM, 0, 0, + &hpo_enable_control, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO", "Switch", "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0100); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL, 0x4, 0x4); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x8); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + regmap_write(rt286->regmap, RT286_SET_AUDIO_D0, 0); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + regmap_write(rt286->regmap, + RT286_SET_D3(rt286_support_power_controls[i]), 0); + } + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2, 0x0000); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MIC1_DET_CTRL, 0x0816); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, 0x0000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_4000, 0); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_0, 0); + + mdelay(10); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + if (rt286->i2c->irq) { + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_IRQ_CTRL, 0x2, 0x2); + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..5f9dc99 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,159 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_CONNECTION_INDEX_MIC1 0X0 +#define RT286_CONNECTION_INDEX_DMIC 0X5 + +/* Verb commands */ +#define RT286_SET_D0(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D0) +#define RT286_SET_D3(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D3) + +#define RT286_SET_MIC1_24\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x24) +#define RT286_SET_MIC1_20\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x20) +#define RT286_SET_AUDIO_D0 RT286_SET_D0(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_AUDIO_D3 RT286_SET_D3(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_D0 RT286_SET_D0(RT286_HP_OUT) +#define RT286_SET_HPO_D3 RT286_SET_D3(RT286_HP_OUT) +#define RT286_SET_SPK_D0 RT286_SET_D0(RT286_SPK_OUT) +#define RT286_SET_SPK_D3 RT286_SET_D3(RT286_SPK_OUT) +#define RT286_SET_DMIC1_D0 RT286_SET_D0(RT286_DMIC1) +#define RT286_SET_DMIC1_D3 RT286_SET_D3(RT286_DMIC1) +#define RT286_ENABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0x40) +#define RT286_DISABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_ENABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0x40) +#define RT286_DISABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_ENABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0x20) +#define RT286_DISABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD_HIGH\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0x2) +#define RT286_SPK_EAPD_LOW\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_UNMUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb000) +#define RT286_UNMUTE_HPO_R\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_UNMUTE_HPO_L\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_MUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb080) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT_4000\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0x4000) +#define RT286_SET_DMIC2_DEFAULT_0\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

5 14

[alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Arun Shamanna Lakshmi 21 Mar '14

21 Mar '14

Currently soc_enum structure supports only 2 registers (reg, reg2) for kcontrol. However, it is possible to have multiple registers per mux. This change allows us to control these multiple registers. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..ddedfb4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1093,6 +1093,9 @@ struct soc_enum { unsigned int mask; const char * const *texts; const unsigned int *values; + unsigned int *regs; + unsigned int *masks; + unsigned int num_regs; }; /* codec IO */ -- 1.7.9.5

4 8

[alsa-devel] [PATCH 1/2] ASoC: fsl: make sure the regmap-mmio is actually enabled
by Xiubo Li 21 Mar '14

21 Mar '14

Since the Freescale SAI, ESAI and SPDIF are using the regmap-mmio, there needs a select of REGMAP_MMIO in the Kconfig to ensure that regmap-mmio is actually enabled. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Cc: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..d0914c0 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_FSL_SAI tristate + select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_FSL_SSI @@ -7,9 +8,11 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_ESAI tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -204,7 +207,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- 1.8.4

4 7

[alsa-devel] [PATCH v4] ASoC: cs42xx8: Add codec driver support for CS42448/CS42888
by Nicolin Chen 20 Mar '14

20 Mar '14

This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that has six/four 24-bit AD and eight 24-bit DA converters. [ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- Changelog v4: * Added CS42448 supports * Renamed the driver to a gernal one -- cs42xx8 v3: * Refined document at 'clocks' section. v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42xx8.txt | 28 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42xx8-i2c.c | 64 +++ sound/soc/codecs/cs42xx8.c | 602 +++++++++++++++++++++ sound/soc/codecs/cs42xx8.h | 238 ++++++++ 6 files changed, 946 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt new file mode 100644 index 0000000..f631fbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -0,0 +1,28 @@ +CS42448/CS42888 audio CODEC + +Required properties: + + - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1a8ff1e..f0e8401 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -304,6 +305,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42XX8 + tristate + +config SND_SOC_CS42XX8_I2C + tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42XX8 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73df822..3c4d275 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42xx8-objs := cs42xx8.o +snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -179,6 +181,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o +obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c new file mode 100644 index 0000000..657dce2 --- /dev/null +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -0,0 +1,64 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> + +#include "cs42xx8.h" + +static int cs42xx8_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42xx8_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42xx8_i2c_id[] = { + {"cs42448", (kernel_ulong_t)&cs42448_data}, + {"cs42888", (kernel_ulong_t)&cs42888_data}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); + +static struct i2c_driver cs42xx8_i2c_driver = { + .driver = { + .name = "cs42xx8", + .owner = THIS_MODULE, + .pm = &cs42xx8_pm, + }, + .probe = cs42xx8_i2c_probe, + .remove = cs42xx8_i2c_remove, + .id_table = cs42xx8_i2c_id, +}; + +module_i2c_driver(cs42xx8_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c new file mode 100644 index 0000000..082299a --- /dev/null +++ b/sound/soc/codecs/cs42xx8.c @@ -0,0 +1,602 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42xx8.h" + +#define CS42XX8_NUM_SUPPLIES 4 +static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42xx8_priv { + struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES]; + const struct cs42xx8_driver_data *drvdata; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single); +static const struct soc_enum adc3_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc); + +static const struct snd_kcontrol_new cs42xx8_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1, + CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3, + CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5, + CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7, + CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1, + CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3, + CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5, + CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0), + SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum), +}; + +static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1), + + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), +}; + +static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { + /* Capture */ + { "ADC3", NULL, "AIN3L" }, + { "ADC3", NULL, "AIN3R" }, + { "ADC3", NULL, "PWR" }, +}; + +struct cs42xx8_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42xx8_ratios cs42xx8_ratios[] = { + { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, + { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, + { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, + { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, + { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, + { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, + { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, + { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + cs42xx8->sysclk = freq; + + return 0; +} + +static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF, + CS42XX8_INTF_DAC_DIF_MASK | + CS42XX8_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42xx8->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42xx8->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42xx8_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42xx8->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { + if (cs42xx8_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42xx8_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mclk; + + fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, + CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42xx8_dai_ops = { + .set_fmt = cs42xx8_set_dai_fmt, + .set_sysclk = cs42xx8_set_dai_sysclk, + .hw_params = cs42xx8_hw_params, + .digital_mute = cs42xx8_digital_mute, +}; + +static struct snd_soc_dai_driver cs42xx8_dai = { + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .ops = &cs42xx8_dai_ops, +}; + +static const struct reg_default cs42xx8_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x15, 0x00 }, /* Volume Control AIN5 */ + { 0x16, 0x00 }, /* Volume Control AIN6 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_STATUS: + return true; + default: + return false; + } +} + +static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_CHIPID: + case CS42XX8_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42xx8_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42XX8_LASTREG, + .reg_defaults = cs42xx8_reg, + .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg), + .volatile_reg = cs42xx8_volatile_register, + .writeable_reg = cs42xx8_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); + +static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + switch (cs42xx8->drvdata->num_adcs) { + case 3: + snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + ARRAY_SIZE(cs42xx8_adc3_snd_controls)); + snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, + ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes, + ARRAY_SIZE(cs42xx8_adc3_dapm_routes)); + break; + default: + break; + } + + /* Mute all DAC channels */ + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42xx8_driver = { + .probe = cs42xx8_codec_probe, + .idle_bias_off = true, + + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), +}; + +const struct cs42xx8_driver_data cs42448_data = { + .name = "cs42448", + .num_adcs = 3, +}; +EXPORT_SYMBOL_GPL(cs42448_data); + +const struct cs42xx8_driver_data cs42888_data = { + .name = "cs42888", + .num_adcs = 2, +}; +EXPORT_SYMBOL_GPL(cs42888_data); + +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); +EXPORT_SYMBOL_GPL(cs42xx8_of_match); + +int cs42xx8_probe(struct device *dev, struct regmap *regmap) +{ + const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev); + struct cs42xx8_priv *cs42xx8; + int ret, val, i; + + cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL); + if (cs42xx8 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42xx8); + + if (of_id) + cs42xx8->drvdata = of_id->data; + + if (!cs42xx8->drvdata) { + dev_err(dev, "failed to find driver data\n"); + return -EINVAL; + } + + cs42xx8->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42xx8->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42xx8->clk)); + return -EINVAL; + } + + cs42xx8->sysclk = clk_get_rate(cs42xx8->clk); + + for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++) + cs42xx8->supplies[i].supply = cs42xx8_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42xx8->regmap = regmap; + if (IS_ERR(cs42xx8->regmap)) { + ret = PTR_ERR(cs42xx8->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42xx8->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42XX8_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42XX8_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42xx8->regmap, false); + + cs42xx8_dai.name = cs42xx8->drvdata->name; + + /* Each adc supports stereo input */ + cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; + + ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42xx8->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42xx8_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42xx8_runtime_resume(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42xx8->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42xx8->regmap, false); + + ret = regcache_sync(cs42xx8->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); +err_clk: + clk_disable_unprepare(cs42xx8->clk); + + return ret; +} + +static int cs42xx8_runtime_suspend(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + + regcache_cache_only(cs42xx8->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + clk_disable_unprepare(cs42xx8->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42xx8_pm = { + SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42xx8_pm); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h new file mode 100644 index 0000000..da0b94a --- /dev/null +++ b/sound/soc/codecs/cs42xx8.h @@ -0,0 +1,238 @@ +/* + * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42XX8_H +#define _CS42XX8_H + +struct cs42xx8_driver_data { + char name[32]; + int num_adcs; +}; + +extern const struct dev_pm_ops cs42xx8_pm; +extern const struct cs42xx8_driver_data cs42448_data; +extern const struct cs42xx8_driver_data cs42888_data; +extern const struct regmap_config cs42xx8_regmap_config; +int cs42xx8_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42XX8_CHIPID 0x01 /* Chip ID */ +#define CS42XX8_PWRCTL 0x02 /* Power Control */ +#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */ +#define CS42XX8_INTF 0x04 /* Interface Formats */ +#define CS42XX8_ADCCTL 0x05 /* ADC Control */ +#define CS42XX8_TXCTL 0x06 /* Transition Control */ +#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */ +#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */ +#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */ +#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42XX8_STATUSCTL 0x18 /* Status Control */ +#define CS42XX8_STATUS 0x19 /* Status */ +#define CS42XX8_STATUSM 0x1A /* Status Mask */ +#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42XX8_FIRSTREG CS42XX8_CHIPID +#define CS42XX8_LASTREG CS42XX8_MUTEC +#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1) +#define CS42XX8_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42XX8_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7 +#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_SHIFT 0 +#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT) +#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK) +#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v)) +#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1 +#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3 +#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT) +#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42XX8_FM_SINGLE 0 +#define CS42XX8_FM_DOUBLE 1 +#define CS42XX8_FM_QUAD 2 +#define CS42XX8_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42XX8_INTF_FREEZE_SHIFT 7 +#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_AUX_DIF_SHIFT 6 +#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_SHIFT 3 +#define CS42XX8_INTF_DAC_DIF_WIDTH 3 +#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_SHIFT 0 +#define CS42XX8_INTF_ADC_DIF_WIDTH 3 +#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2 +#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1 +#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0 +#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5 +#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2 +#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_AMUTE_SHIFT 4 +#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0 +#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n) +#define CS42XX8_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42XX8_STATUSCTL_INI_SHIFT 2 +#define CS42XX8_STATUSCTL_INI_WIDTH 2 +#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42XX8_H */ -- 1.8.4

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[alsa-devel] [PATCH v3 0/4] ASoC: simple-card: multi DAI links extension
by Jean-Francois Moine 20 Mar '14

20 Mar '14

This patch series extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. -v3 - remove 'Fix the reference count of device nodes' which is applied (Mark Brown) - new patch 'Simplify code' - dynamically allocate and use properties for all DAI links (Jyri Sarha and Li Xiubo) - v2 - change subject/comment about device node reference count (Mark Brown) - use a null size array instead of an implicit area for the DAI links (Li Xiubo) - update the reference count of the device node at end of probe Jean-Francois Moine (4): ASoC: simple-card: Simplify code ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Handle many DAI links ASoC: simple-card: Add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 +++- sound/soc/generic/simple-card.c | 181 +++++++++++++-------- 2 files changed, 145 insertions(+), 70 deletions(-) -- 1.9.0

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