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

[alsa-devel] [patch] ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
by Dan Carpenter 29 May '14

29 May '14

ARRAY_SIZE() was intended here instead of sizeof(). The "bridgeco_freq_table" array holds integers so the original condition is never true. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 514c7c9..bc4f827 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -749,7 +749,7 @@ parse_stream_formation(u8 *buf, unsigned int len, if (buf[2] == bridgeco_freq_table[i]) break; } - if (i == sizeof(bridgeco_freq_table)) + if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */

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[alsa-devel] [patch 2/2] ALSA: fireworks: small leak on error path
by Dan Carpenter 29 May '14

29 May '14

There was a typo here so we return directly instead of freeing "hwinfo". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8e856cc..996fdc4 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -157,7 +157,7 @@ get_hardware_info(struct snd_efw *efw) /* Hardware metering. */ if (hwinfo->phys_in_grp_count > HWINFO_MAX_CAPS_GROUPS || hwinfo->phys_out_grp_count > HWINFO_MAX_CAPS_GROUPS) { - return -EIO; + err = -EIO; goto end; } efw->phys_in = hwinfo->phys_in;

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[alsa-devel] [patch 1/2] ALSA: fireworks: remove some stray checks
by Dan Carpenter 29 May '14

29 May '14

We checked "err" earlier. These things seem to be left over code. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index a354d26..8e856cc 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -89,8 +89,6 @@ get_hardware_info(struct snd_efw *efw) snprintf(version, sizeof(version), "%u.%u", (hwinfo->arm_version >> 24) & 0xff, (hwinfo->arm_version >> 16) & 0xff); - if (err < 0) - goto end; efw->firmware_version = hwinfo->arm_version; strcpy(efw->card->driver, "Fireworks"); @@ -101,8 +99,6 @@ get_hardware_info(struct snd_efw *efw) hwinfo->vendor_name, hwinfo->model_name, version, hwinfo->guid_hi, hwinfo->guid_lo, dev_name(&efw->unit->device), 100 << fw_dev->max_speed); - if (err < 0) - goto end; if (hwinfo->flags & BIT(FLAG_RESP_ADDR_CHANGABLE)) efw->resp_addr_changable = true;

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[alsa-devel] [PATCH 1/2] ASoC: export dapm_kcontrol_set/get_value functions
by Vinod Koul 29 May '14

29 May '14

The DSPs like Intel ones use the DPCM to represent the DSP toplogy. So when DAPM triggers the widget, the DSP needs to know the kcontrol values and pass on to DSP firmware, so export these for driver use Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 6 ++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..abda467 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,6 +368,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, + unsigned int value); +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 98c1dc6..d831e73 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -327,14 +327,15 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); -static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, +bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); @@ -349,6 +350,7 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_set_value); /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol -- 1.7.0.4

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[alsa-devel] [PATCH 0/2] Use resource_size instead of computation
by Benoit Taine 28 May '14

28 May '14

These patches enhance kernel style usage, and make use of the resource_size() function instead of explicit computation. The semantic patch at scripts/coccinelle/api/resource_size.cocci was used to detect and edit this situation.

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[alsa-devel] [RFC] AVB - network-based soundcards in ALSA
by Henrik Austad 28 May '14

28 May '14

Hi all! This is an RFC for a new class of soundcards. I am not very familiar with how ALSA is tied together underneath the hood, so what you see here, is based on my naive understanding of ALSA. I wear asbestos underwear on a regular basis, so I prefer honesty over sugarcoating :) I use "I" and "we" interchangeably. By 'we' I mean a small R&D group at Cisco Norway, by "I", I mean.. well, me. So, we plan for AVB, I do the kernel side work. We plan to upstream this, given that the community accepts it. Also, I've used my private address as that is set up to track kernel-related lists, but added my Cisco-address so please keep that on the CC if you reply. We have recently begun working on Audio Video Bridging (AVB, [1]) and is looking into how this can be added to the Linux Kernel via ALSA and video4linux. But first; for those of you who are not familiar with AVB: In short, AVB is just a set of open standards governing network and timing configuration so that you can stream audio and video reliably and with low latency. Note that this is not the kind of streaming services currently associated with streaming (a few companies distributing movies and TV-shows comes to mind; one rhyming with lightsticks). It is the kind of streaming you use when connecting a pair of speakers to your computer - via ethernet. Or a webcam via the wireless network. (I'm aware of the security implications here, but bear with me). For the eager reader, AVB is being promoted by AVnu Alliance [2], they have a lot of information available. I also added a link to a very short intro to AVB that Hans held a few weeks back (focus on the network though) in [3]. Then the IEEE 802.1 working group [4] has a few standards, but these are probably not that relevant no this list, at least not right now. For AVB to work, you need support in the networking infrastructure. This is not prevalent but it is coming. There are a few manufacturers that provide AVB ready equipment and some networking gear. What you need of standards for AVB: * gPTP support (IEEE 802.1AS), this is an IEEE 1588 (PTP) profile for AVB. This is needed for accurate timestamping of samples, and all nodes in an AVB domain must agree to the _same_ time (not that the _correct_ time is not that important in this setting). .1AS should give you a <1us error between the clocks for the systems involved. * Stream Reservation (IEEE 802.1Qat, or 802.1Q:2011 Sec. #35) to make sure we have guaranteed bandwidth. This will avoid dropped etherframes due to congested network. It also caps the amount you can reserve to 75% of total BW, making sure AVB can coexist with normal traffic. * Traffic Shaping and adminssion control (IEEE 802.1Qav, or 802.1Q:2011 Sec. #34) to improve utilization but also avoid/minimize jitter due to queues inside switches/routers/bridges. * IEEE 802.1BA, default configuration for AVB devices and what the network looks like. * IEEE 1722 (and 1733 for layer-3) Layer 2 Transport for audio/video. The packing is similar to what is done in Firewire. You have 8kHz frame intervals for class A, 4kHz for class B. This gives relatively few samples pr. frame. Currently we only look at Layer 2 as small peripherals (microphones, speakers) will only have to implement L2 instead of the entire IP-stack. * IEEE 1722.1 Device discovery protocol (AVDECC) defines how Talkers and Listeners find each other and connect. Any talker will regularly announce its presence, and 1722.1 defines how to announce - and how to respond. Of all these standards, the 802.1BA and 1722 are probably the most interesting ones. AVnu also has a 'best practice' [5] document that gives a outline that serves as a nice starting point. Terminology (brief) - Bridge: Node in the network with more than 1 port (think switches) - End-station: Node in the network with 1 port. - Talkers: End-station that produce media (mic, camera) - Listeners: End-station that receives from Talkers - Streams & Channels: A talker creates a stream through the network to a Listener. Each stream is composed of 1..N channels where each sample is interleaved. - An end-station can act as both Talker and Listener. - gPTP domain: set of PTP-capable nodes connected (gPTP will not allow non-timeaware nodes in the domain). - SRP domain: nodes in a network that supports stream reservation. - AVB Domain: intersection of SRP domain and gPTP domain. To put it to easier terms, AVB gives you a way to add 'stuff' to your computer and play music to them via the network. Moving out into ALSA-land and introducing "The plan": * A central driver, an "avb_core" if you like. Once loaded it will create a configfs directory and start looking at etherframes to see if anything of interest comes along. This will be present from the start and is required for all the rest to work. * An "avb_media_driver" to split data going to ALSA and v4l as well as combining streams coming back. The easiest way is probably to combine snd_avb and the corresponding v4l driver into a single driver, but expose it as "snd_avb" to ALSA (and ditto for v4l2). * A userspace tool for tapping into the AVDECC data (for autodiscovery of nodes). Let's call this avdecclib for now (there are a few userspace libraries available on github). * ConfigFS [6] is then used by userspace to spawn an new avb_media_driver for each stream we want to connect to. Tree-structure will look something like this mkdir /config/avb/node0; config/ └── avb └── node0 ├── channels_in ├── channels_out ├── enable └── mac (the number of attributes will have to be adjusted as I figure out what makes sense to have in the configfs item. Writing 1 to enable will then trigger the negotiating phase and wait for the driver to come online. A new ALSA soundcard will then pop into existence, which can then be used as any regular soundcard attached to the computer. So, an attempt to bring this to life using state of the art ASCII skills +----------------------------------------------------+ | | | media application | | | +-------+-----------------+--------------------+-----+ | | | | | | +-------+-----+ +------+------+ +-------+-----+ | | | | | | | alsalib | | v4l2lib | | avdecclib | | | | | | | userspace +-------+--- -+ +------+------+ +-------+-----+ ................ | | | kernelspace | | | +-------+-----+ | | | | | | | alsa core | | | | | | | +-------+-----+ | | | | | +-------+-----------------+------+ | | | +-------+-----+ | snd_avb v4l2_avb | | | | | | ConfigFS | | avb_media_driver | | | | | +-------+-----+ +-------------+------------+-----+ | | | +-------+-----+ +---------------+ +------+------+ | | | | +----------+ +-----------------| avb_config | | time | | avb_core | | | | | +-----+ | | | +-------------+ +-------+-------+ | +------+------+ | | | | | | | | | +-------+-------+ | | | | | | | | | media_clock | | +------+------+ | | | | | | | +---------------+ +----+ net +-----+ | | +-------------+ So, why in the kernel and not completely in userspace? Primarily because we would like to make it as easy as possible to create a Talker or a Listener in an AVB domain. Sure, you would need some kind of tool to manage the ConfigFS interface and set up the detailed configuration, but once that is done, _any_ program on a standard GNU/Linux box can use AVB as if it was a regular soundcard. That is a real benefit, and what makes it really exciting. It is also a bit difficult to associate a physical location to a MAC-address. A userspace tool can be configured to remember this, but this is not information that belongs in the kernel. This needs to be persistent anyway, so setting 00:00:A4.. to be "L&R Speaker in Henrik's Den" doesn't really make sense to compile into the kernel. Then there is the notion of security. If the kernel triggers on every newly discovered device, it is pretty simple to write a metasploit plugin that will bring any AVB enabled Linux box to its knees by just flooding the network with Announce-messages. Also, I don't necessarily want the stream from my computer to my speakers to be accessed by someone (tm) on my network. I'd greatly appreciate feedback and comments, especially with regards to the rough outline and the usage of ConfigFS and ioctls. Stay tuned! Once we have something that doesn't crash and burn in the most horrible sense, I'll submit a few patches for people to look at. If the interest is high, I'll probably create a public repo that I'll update more frequently, but with more of the bleeding-part of the edge. Thanks! 1) http://en.wikipedia.org/wiki/Audio_Video_Bridging 2) http://www.avnu.org/ 3) http://www.slideshare.net/henrikau/avb-v4l2summit 4) http://en.wikipedia.org/wiki/IEEE_802.1 5) http://www.avnu.org/knowledge_center 6) http://events.linuxfoundation.org/sites/events/files/slides/USB%20Gadget%20… -- Henrik Austad

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[alsa-devel] [PATCH 0/2] ASoC: tlv320aic3x: Fix snd_soc_dapm_put_volsw_aic3x callback
by Peter Ujfalusi 28 May '14

28 May '14

Hi, Commit "cf7c1de20c576 ASoC: dapm: Move 'value' field from widget to control" changed the way how the 'value' has been stored for a widget. This resulted issues with codec drivers needing and using custom dapm put callbacks to handle the hw underneath, like the tlv320aic3x driver. Since the driver is not updated following the mentioned commit it is mostly broken when we try to change Left/Righ PGA mixers: amixer sset 'Left PGA Mixer Line1L' off amixer: Invalid command! The reason for this is that the custom callback was not calling the dapm_kcontrol_set_value() to store the value which causes inconsistent state between the HW and dapm. Regards, Peter --- Peter Ujfalusi (2): ASoC: dapm: Export dapm_kcontrol_set_value() so it can be used by drivers ASoC: tlv320aic3x: Fix Left/Rigth PGA mixer widgets/kcontrols include/sound/soc-dapm.h | 3 +++ sound/soc/codecs/tlv320aic3x.c | 14 +++++++------- sound/soc/soc-dapm.c | 3 ++- 3 files changed, 12 insertions(+), 8 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH 1/5] ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
by Lars-Peter Clausen 28 May '14

28 May '14

The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite different in the analog domain. This patch adds support for the common parts of the ADAU1X61 and ADAU1X81 CODECs. The patch also restores some of the alphabetical order in the Makfile and Kconfig. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/platform_data/adau17x1.h | 23 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/adau17x1.c | 866 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau17x1.h | 124 +++++ 5 files changed, 1022 insertions(+), 3 deletions(-) create mode 100644 include/linux/platform_data/adau17x1.h create mode 100644 sound/soc/codecs/adau17x1.c create mode 100644 sound/soc/codecs/adau17x1.h diff --git a/include/linux/platform_data/adau17x1.h b/include/linux/platform_data/adau17x1.h new file mode 100644 index 0000000..f90bd92 --- /dev/null +++ b/include/linux/platform_data/adau17x1.h @@ -0,0 +1,23 @@ +/* + * Driver for ADAU1761/ADAU1461/ADAU1761/ADAU1961/ADAU1781/ADAU1781 codecs + * + * Copyright 2011-2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU17X1_H__ +#define __LINUX_PLATFORM_DATA_ADAU17X1_H__ + +/** + * enum adau17x1_micbias_voltage - Microphone bias voltage + * @ADAU17X1_MICBIAS_0_90_AVDD: 0.9 * AVDD + * @ADAU17X1_MICBIAS_0_65_AVDD: 0.65 * AVDD + */ +enum adau17x1_micbias_voltage { + ADAU17X1_MICBIAS_0_90_AVDD = 0, + ADAU17X1_MICBIAS_0_65_AVDD = 1, +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3f7bbe6..1f308a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -214,13 +214,17 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C select SND_SOC_SIGMADSP -config SND_SOC_ADAU1373 +config SND_SOC_ADAU17X1 tristate + select SND_SOC_SIGMADSP config SND_SOC_ADAU1977 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1ccdaf0..ee04f5a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,8 +7,9 @@ snd-soc-ad193x-spi-objs := ad193x-spi.o snd-soc-ad193x-i2c-objs := ad193x-i2c.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o -snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau1701-objs := adau1701.o +snd-soc-adau17x1-objs := adau17x1.o snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o @@ -162,10 +163,11 @@ obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o -obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c new file mode 100644 index 0000000..2961fae --- /dev/null +++ b/sound/soc/codecs/adau17x1.c @@ -0,0 +1,866 @@ +/* + * Common code for ADAU1X61 and ADAU1X81 codecs + * + * Copyright 2011-2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/gcd.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include "sigmadsp.h" +#include "adau17x1.h" + +static const char * const adau17x1_capture_mixer_boost_text[] = { + "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_capture_boost_enum, + ADAU17X1_REC_POWER_MGMT, 5, adau17x1_capture_mixer_boost_text); + +static const char * const adau17x1_mic_bias_mode_text[] = { + "Normal operation", "High performance", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_mic_bias_mode_enum, + ADAU17X1_MICBIAS, 3, adau17x1_mic_bias_mode_text); + +static const DECLARE_TLV_DB_MINMAX(adau17x1_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adau17x1_controls[] = { + SOC_DOUBLE_R_TLV("Digital Capture Volume", + ADAU17X1_LEFT_INPUT_DIGITAL_VOL, + ADAU17X1_RIGHT_INPUT_DIGITAL_VOL, + 0, 0xff, 1, adau17x1_digital_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", ADAU17X1_DAC_CONTROL1, + ADAU17X1_DAC_CONTROL2, 0, 0xff, 1, adau17x1_digital_tlv), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAU17X1_ADC_CONTROL, + 5, 1, 0), + SOC_SINGLE("Playback De-emphasis Switch", ADAU17X1_DAC_CONTROL0, + 2, 1, 0), + + SOC_ENUM("Capture Boost", adau17x1_capture_boost_enum), + + SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), +}; + +static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + int ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + adau->pll_regs[5] = 1; + } else { + adau->pll_regs[5] = 0; + /* Bypass the PLL when disabled, otherwise registers will become + * inaccessible. */ + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, 0); + } + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + mdelay(5); + regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL, + ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL); + } + + return 0; +} + +static const char * const adau17x1_mono_stereo_text[] = { + "Stereo", + "Mono Left Channel (L+R)", + "Mono Right Channel (L+R)", + "Mono (L+R)", +}; + +static SOC_ENUM_SINGLE_DECL(adau17x1_dac_mode_enum, + ADAU17X1_DAC_CONTROL0, 6, adau17x1_mono_stereo_text); + +static const struct snd_kcontrol_new adau17x1_dac_mode_mux = + SOC_DAPM_ENUM("DAC Mono-Stereo-Mode", adau17x1_dac_mode_enum); + +static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("PLL", 3, SND_SOC_NOPM, 0, 0, adau17x1_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("AIFCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU17X1_MICBIAS, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Left Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Right Playback Enable", ADAU17X1_PLAY_POWER_MGMT, + 1, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Left DAC Mode Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mode_mux), + SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mode_mux), + + SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0), + SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_routes[] = { + { "Left Decimator", NULL, "SYSCLK" }, + { "Right Decimator", NULL, "SYSCLK" }, + { "Left DAC", NULL, "SYSCLK" }, + { "Right DAC", NULL, "SYSCLK" }, + { "Capture", NULL, "SYSCLK" }, + { "Playback", NULL, "SYSCLK" }, + + { "Left DAC", NULL, "Left DAC Mode Mux" }, + { "Right DAC", NULL, "Right DAC Mode Mux" }, + + { "Capture", NULL, "AIFCLK" }, + { "Playback", NULL, "AIFCLK" }, +}; + +static const struct snd_soc_dapm_route adau17x1_dapm_pll_route = { + "SYSCLK", NULL, "PLL", +}; + +/* + * The MUX register for the Capture and Playback MUXs selects either DSP as + * source/destination or one of the TDM slots. The TDM slot is selected via + * snd_soc_dai_set_tdm_slot(), so we only expose whether to go to the DSP or + * directly to the DAI interface with this control. + */ +static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int stream = e->shift_l; + unsigned int val, change; + int reg; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + val = 0; + adau->dsp_bypass[stream] = false; + break; + default: + val = (adau->tdm_slot[stream] * 2) + 1; + adau->dsp_bypass[stream] = true; + break; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ADAU17X1_SERIAL_INPUT_ROUTE; + else + reg = ADAU17X1_SERIAL_OUTPUT_ROUTE; + + change = snd_soc_test_bits(codec, reg, 0xff, val); + if (change) { + update.kcontrol = kcontrol; + update.mask = 0xff; + update.reg = reg; + update.val = val; + + snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol, + ucontrol->value.enumerated.item[0], e, &update); + } + + return change; +} + +static int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct adau *adau = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int stream = e->shift_l; + unsigned int reg, val; + int ret; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ADAU17X1_SERIAL_INPUT_ROUTE; + else + reg = ADAU17X1_SERIAL_OUTPUT_ROUTE; + + ret = regmap_read(adau->regmap, reg, &val); + if (ret) + return ret; + + if (val != 0) + val = 1; + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +#define DECLARE_ADAU17X1_DSP_MUX_CTRL(_name, _label, _stream, _text) \ + const struct snd_kcontrol_new _name = \ + SOC_DAPM_ENUM_EXT(_label, (const struct soc_enum)\ + SOC_ENUM_SINGLE(SND_SOC_NOPM, _stream, \ + ARRAY_SIZE(_text), _text), \ + adau17x1_dsp_mux_enum_get, adau17x1_dsp_mux_enum_put) + +static const char * const adau17x1_dac_mux_text[] = { + "DSP", + "AIFIN", +}; + +static const char * const adau17x1_capture_mux_text[] = { + "DSP", + "Decimator", +}; + +static DECLARE_ADAU17X1_DSP_MUX_CTRL(adau17x1_dac_mux, "DAC Playback Mux", + SNDRV_PCM_STREAM_PLAYBACK, adau17x1_dac_mux_text); + +static DECLARE_ADAU17X1_DSP_MUX_CTRL(adau17x1_capture_mux, "Capture Mux", + SNDRV_PCM_STREAM_CAPTURE, adau17x1_capture_mux_text); + +static const struct snd_soc_dapm_widget adau17x1_dsp_dapm_widgets[] = { + SND_SOC_DAPM_PGA("DSP", ADAU17X1_DSP_RUN, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("DSP Siggen"), + + SND_SOC_DAPM_MUX("DAC Playback Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_dac_mux), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &adau17x1_capture_mux), +}; + +static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { + { "DAC Playback Mux", "DSP", "DSP" }, + { "DAC Playback Mux", "AIFIN", "Playback" }, + + { "Left DAC Mode Mux", "Stereo", "DAC Playback Mux" }, + { "Left DAC Mode Mux", "Mono (L+R)", "DAC Playback Mux" }, + { "Left DAC Mode Mux", "Mono Left Channel (L+R)", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Stereo", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Mono (L+R)", "DAC Playback Mux" }, + { "Right DAC Mode Mux", "Mono Right Channel (L+R)", "DAC Playback Mux" }, + + { "Capture Mux", "DSP", "DSP" }, + { "Capture Mux", "Decimator", "Left Decimator" }, + { "Capture Mux", "Decimator", "Right Decimator" }, + + { "Capture", NULL, "Capture Mux" }, + + { "DSP", NULL, "DSP Siggen" }, + + { "DSP", NULL, "Left Decimator" }, + { "DSP", NULL, "Right Decimator" }, +}; + +static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { + { "Left DAC Mode Mux", "Stereo", "Playback" }, + { "Left DAC Mode Mux", "Mono (L+R)", "Playback" }, + { "Left DAC Mode Mux", "Mono Left Channel (L+R)", "Playback" }, + { "Right DAC Mode Mux", "Stereo", "Playback" }, + { "Right DAC Mode Mux", "Mono (L+R)", "Playback" }, + { "Right DAC Mode Mux", "Mono Right Channel (L+R)", "Playback" }, + { "Capture", NULL, "Left Decimator" }, + { "Capture", NULL, "Right Decimator" }, +}; + +bool adau17x1_has_dsp(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1381: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_dsp); + +static int adau17x1_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 adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int val, div, dsp_div; + unsigned int freq; + + if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + freq = adau->pll_freq; + else + freq = adau->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* fs */ + div = 0; + dsp_div = 1; + break; + case 6144: /* fs / 6 */ + div = 1; + dsp_div = 6; + break; + case 4096: /* fs / 4 */ + div = 2; + dsp_div = 5; + break; + case 3072: /* fs / 3 */ + div = 3; + dsp_div = 4; + break; + case 2048: /* fs / 2 */ + div = 4; + dsp_div = 3; + break; + case 1536: /* fs / 1.5 */ + div = 5; + dsp_div = 2; + break; + case 512: /* fs / 0.5 */ + div = 6; + dsp_div = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_CONVSR_MASK, div); + if (adau17x1_has_dsp(adau)) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_SAMPLING_RATE, div); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); + } + + if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); +} + +static int adau17x1_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 adau *adau = snd_soc_codec_get_drvdata(codec); + unsigned int r, n, m, i, j; + unsigned int div; + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + if (!freq_out) { + r = 0; + n = 0; + m = 0; + div = 0; + } else { + if (freq_out % freq_in != 0) { + div = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= div; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + div--; + } else { + r = freq_out / freq_in; + n = 0; + m = 0; + div = 0; + } + if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) + return -EINVAL; + } + + adau->pll_regs[0] = m >> 8; + adau->pll_regs[1] = m & 0xff; + adau->pll_regs[2] = n >> 8; + adau->pll_regs[3] = n & 0xff; + adau->pll_regs[4] = (r << 3) | (div << 1); + if (m != 0) + adau->pll_regs[4] |= 1; /* Fractional mode */ + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + struct snd_soc_dapm_context *dapm = &dai->codec->dapm; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + case ADAU17X1_CLK_SRC_PLL: + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (adau->clk_src != clk_id) { + if (clk_id == ADAU17X1_CLK_SRC_PLL) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1; + int lrclk_pol; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl0 = ADAU17X1_SERIAL_PORT0_MASTER; + adau->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl0 = 0; + adau->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + lrclk_pol = 0; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + lrclk_pol = 1; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + case SND_SOC_DAIFMT_DSP_A: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY1; + break; + case SND_SOC_DAIFMT_DSP_B: + lrclk_pol = 1; + ctrl0 |= ADAU17X1_SERIAL_PORT0_PULSE_MODE; + ctrl1 = ADAU17X1_SERIAL_PORT1_DELAY0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk_pol = !lrclk_pol; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl0 |= ADAU17X1_SERIAL_PORT0_BCLK_POL; + lrclk_pol = !lrclk_pol; + break; + default: + return -EINVAL; + } + + if (lrclk_pol) + ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; + + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); + regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + + adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ser_ctrl0, ser_ctrl1; + unsigned int conv_ctrl0, conv_ctrl1; + + /* I2S mode */ + if (slots == 0) { + slots = 2; + rx_mask = 3; + tx_mask = 3; + slot_width = 32; + } + + switch (slots) { + case 2: + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_STEREO; + break; + case 4: + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_TDM4; + break; + case 8: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl0 = ADAU17X1_SERIAL_PORT0_TDM8; + break; + default: + return -EINVAL; + } + + switch (slot_width * slots) { + case 32: + if (adau->type == ADAU1761) + return -EINVAL; + + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK32; + break; + case 64: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK64; + break; + case 48: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK48; + break; + case 128: + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK128; + break; + case 256: + if (adau->type == ADAU1361) + return -EINVAL; + + ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK256; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0x03: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(1); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 0; + break; + case 0x0c: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(2); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 1; + break; + case 0x30: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(3); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 2; + break; + case 0xc0: + conv_ctrl1 = ADAU17X1_CONVERTER1_ADC_PAIR(4); + adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] = 3; + break; + default: + return -EINVAL; + } + + switch (tx_mask) { + case 0x03: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(1); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 0; + break; + case 0x0c: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(2); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 1; + break; + case 0x30: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(3); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 2; + break; + case 0xc0: + conv_ctrl0 = ADAU17X1_CONVERTER0_DAC_PAIR(4); + adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] = 3; + break; + default: + return -EINVAL; + } + + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_DAC_PAIR_MASK, conv_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER1, + ADAU17X1_CONVERTER1_ADC_PAIR_MASK, conv_ctrl1); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, + ADAU17X1_SERIAL_PORT0_TDM_MASK, ser_ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_BCLK_MASK, ser_ctrl1); + + if (!adau17x1_has_dsp(adau)) + return 0; + + if (adau->dsp_bypass[SNDRV_PCM_STREAM_PLAYBACK]) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_INPUT_ROUTE, + (adau->tdm_slot[SNDRV_PCM_STREAM_PLAYBACK] * 2) + 1); + } + + if (adau->dsp_bypass[SNDRV_PCM_STREAM_CAPTURE]) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_OUTPUT_ROUTE, + (adau->tdm_slot[SNDRV_PCM_STREAM_CAPTURE] * 2) + 1); + } + + return 0; +} + +const struct snd_soc_dai_ops adau17x1_dai_ops = { + .hw_params = adau17x1_hw_params, + .set_sysclk = adau17x1_set_dai_sysclk, + .set_fmt = adau17x1_set_dai_fmt, + .set_pll = adau17x1_set_dai_pll, + .set_tdm_slot = adau17x1_set_dai_tdm_slot, +}; +EXPORT_SYMBOL_GPL(adau17x1_dai_ops); + +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + switch (micbias) { + case ADAU17X1_MICBIAS_0_90_AVDD: + case ADAU17X1_MICBIAS_0_65_AVDD: + break; + default: + return -EINVAL; + } + + return regmap_write(adau->regmap, ADAU17X1_MICBIAS, micbias << 2); +} +EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); + +bool adau17x1_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU17X1_CLOCK_CONTROL: + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_REC_POWER_MGMT: + case ADAU17X1_MICBIAS: + case ADAU17X1_SERIAL_PORT0: + case ADAU17X1_SERIAL_PORT1: + case ADAU17X1_CONVERTER0: + case ADAU17X1_CONVERTER1: + case ADAU17X1_LEFT_INPUT_DIGITAL_VOL: + case ADAU17X1_RIGHT_INPUT_DIGITAL_VOL: + case ADAU17X1_ADC_CONTROL: + case ADAU17X1_PLAY_POWER_MGMT: + case ADAU17X1_DAC_CONTROL0: + case ADAU17X1_DAC_CONTROL1: + case ADAU17X1_DAC_CONTROL2: + case ADAU17X1_SERIAL_PORT_PAD: + case ADAU17X1_CONTROL_PORT_PAD0: + case ADAU17X1_CONTROL_PORT_PAD1: + case ADAU17X1_DSP_SAMPLING_RATE: + case ADAU17X1_SERIAL_INPUT_ROUTE: + case ADAU17X1_SERIAL_OUTPUT_ROUTE: + case ADAU17X1_DSP_ENABLE: + case ADAU17X1_DSP_RUN: + case ADAU17X1_SERIAL_SAMPLING_RATE: + return true; + default: + break; + } + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_readable_register); + +bool adau17x1_volatile_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter and program memory */ + if (reg < 0x4000) + return true; + + switch (reg) { + /* The PLL register is 6 bytes long */ + case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: + return true; + default: + break; + } + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_volatile_register); + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware) +{ + int ret; + int dspsr; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); + if (ret) + return ret; + + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + + ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware); + if (ret) { + regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); + return ret; + } + regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_load_firmware); + +int adau17x1_add_widgets(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_add_codec_controls(codec, adau17x1_controls, + ARRAY_SIZE(adau17x1_controls)); + if (ret) + return ret; + ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets, + ARRAY_SIZE(adau17x1_dapm_widgets)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau17x1_dsp_dapm_widgets, + ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_widgets); + +int adau17x1_add_routes(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes, + ARRAY_SIZE(adau17x1_dapm_routes)); + if (ret) + return ret; + + if (adau17x1_has_dsp(adau)) { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_dsp_dapm_routes)); + } else { + ret = snd_soc_dapm_add_routes(&codec->dapm, + adau17x1_no_dsp_dapm_routes, + ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); + } + return ret; +} +EXPORT_SYMBOL_GPL(adau17x1_add_routes); + +int adau17x1_suspend(struct snd_soc_codec *codec) +{ + codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_suspend); + +int adau17x1_resume(struct snd_soc_codec *codec) +{ + struct adau *adau = snd_soc_codec_get_drvdata(codec); + + if (adau->switch_mode) + adau->switch_mode(codec->dev); + + codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + regcache_sync(adau->regmap); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_resume); + +int adau17x1_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, void (*switch_mode)(struct device *dev)) +{ + struct adau *adau; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau = devm_kzalloc(dev, sizeof(*adau), GFP_KERNEL); + if (!adau) + return -ENOMEM; + + adau->regmap = regmap; + adau->switch_mode = switch_mode; + adau->type = type; + + dev_set_drvdata(dev, adau); + + if (switch_mode) + switch_mode(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(adau17x1_probe); + +MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h new file mode 100644 index 0000000..3ffabaf --- /dev/null +++ b/sound/soc/codecs/adau17x1.h @@ -0,0 +1,124 @@ +#ifndef __ADAU17X1_H__ +#define __ADAU17X1_H__ + +#include <linux/regmap.h> +#include <linux/platform_data/adau17x1.h> + +enum adau17x1_type { + ADAU1361, + ADAU1761, + ADAU1381, + ADAU1781, +}; + +enum adau17x1_pll { + ADAU17X1_PLL, +}; + +enum adau17x1_pll_src { + ADAU17X1_PLL_SRC_MCLK, +}; + +enum adau17x1_clk_src { + ADAU17X1_CLK_SRC_MCLK, + ADAU17X1_CLK_SRC_PLL, +}; + +struct adau { + unsigned int sysclk; + unsigned int pll_freq; + + enum adau17x1_clk_src clk_src; + enum adau17x1_type type; + void (*switch_mode)(struct device *dev); + + unsigned int dai_fmt; + + uint8_t pll_regs[6]; + + bool master; + + unsigned int tdm_slot[2]; + bool dsp_bypass[2]; + + struct regmap *regmap; +}; + +int adau17x1_add_widgets(struct snd_soc_codec *codec); +int adau17x1_add_routes(struct snd_soc_codec *codec); +int adau17x1_probe(struct device *dev, struct regmap *regmap, + enum adau17x1_type type, void (*switch_mode)(struct device *dev)); +int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, + enum adau17x1_micbias_voltage micbias); +bool adau17x1_readable_register(struct device *dev, unsigned int reg); +bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +int adau17x1_suspend(struct snd_soc_codec *codec); +int adau17x1_resume(struct snd_soc_codec *codec); + +extern const struct snd_soc_dai_ops adau17x1_dai_ops; + +int adau17x1_load_firmware(struct adau *adau, struct device *dev, + const char *firmware); +bool adau17x1_has_dsp(struct adau *adau); + +#define ADAU17X1_CLOCK_CONTROL 0x4000 +#define ADAU17X1_PLL_CONTROL 0x4002 +#define ADAU17X1_REC_POWER_MGMT 0x4009 +#define ADAU17X1_MICBIAS 0x4010 +#define ADAU17X1_SERIAL_PORT0 0x4015 +#define ADAU17X1_SERIAL_PORT1 0x4016 +#define ADAU17X1_CONVERTER0 0x4017 +#define ADAU17X1_CONVERTER1 0x4018 +#define ADAU17X1_LEFT_INPUT_DIGITAL_VOL 0x401a +#define ADAU17X1_RIGHT_INPUT_DIGITAL_VOL 0x401b +#define ADAU17X1_ADC_CONTROL 0x4019 +#define ADAU17X1_PLAY_POWER_MGMT 0x4029 +#define ADAU17X1_DAC_CONTROL0 0x402a +#define ADAU17X1_DAC_CONTROL1 0x402b +#define ADAU17X1_DAC_CONTROL2 0x402c +#define ADAU17X1_SERIAL_PORT_PAD 0x402d +#define ADAU17X1_CONTROL_PORT_PAD0 0x402f +#define ADAU17X1_CONTROL_PORT_PAD1 0x4030 +#define ADAU17X1_DSP_SAMPLING_RATE 0x40eb +#define ADAU17X1_SERIAL_INPUT_ROUTE 0x40f2 +#define ADAU17X1_SERIAL_OUTPUT_ROUTE 0x40f3 +#define ADAU17X1_DSP_ENABLE 0x40f5 +#define ADAU17X1_DSP_RUN 0x40f6 +#define ADAU17X1_SERIAL_SAMPLING_RATE 0x40f8 + +#define ADAU17X1_SERIAL_PORT0_BCLK_POL BIT(4) +#define ADAU17X1_SERIAL_PORT0_LRCLK_POL BIT(3) +#define ADAU17X1_SERIAL_PORT0_MASTER BIT(0) + +#define ADAU17X1_SERIAL_PORT1_DELAY1 0x00 +#define ADAU17X1_SERIAL_PORT1_DELAY0 0x01 +#define ADAU17X1_SERIAL_PORT1_DELAY8 0x02 +#define ADAU17X1_SERIAL_PORT1_DELAY16 0x03 +#define ADAU17X1_SERIAL_PORT1_DELAY_MASK 0x03 + +#define ADAU17X1_CLOCK_CONTROL_INFREQ_MASK 0x6 +#define ADAU17X1_CLOCK_CONTROL_CORECLK_SRC_PLL BIT(3) +#define ADAU17X1_CLOCK_CONTROL_SYSCLK_EN BIT(0) + +#define ADAU17X1_SERIAL_PORT1_BCLK32 (0x0 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK48 (0x1 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK64 (0x2 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK128 (0x3 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK256 (0x4 << 5) +#define ADAU17X1_SERIAL_PORT1_BCLK_MASK (0x7 << 5) + +#define ADAU17X1_SERIAL_PORT0_STEREO (0x0 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM4 (0x1 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM8 (0x2 << 1) +#define ADAU17X1_SERIAL_PORT0_TDM_MASK (0x3 << 1) +#define ADAU17X1_SERIAL_PORT0_PULSE_MODE BIT(5) + +#define ADAU17X1_CONVERTER0_DAC_PAIR(x) (((x) - 1) << 5) +#define ADAU17X1_CONVERTER0_DAC_PAIR_MASK (0x3 << 5) +#define ADAU17X1_CONVERTER1_ADC_PAIR(x) ((x) - 1) +#define ADAU17X1_CONVERTER1_ADC_PAIR_MASK 0x3 + +#define ADAU17X1_CONVERTER0_CONVSR_MASK 0x7 + + +#endif -- 1.8.0

3 6

[alsa-devel] [PATCH RFC 02/11] sound: ASoC: kirkwood: Remove unused drivers
by Andrew Lunn 28 May '14

28 May '14

Both kirkwood-openrd and kirkwood-t5325 drivers have been replaced with DT based simple-card equivelents. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/kirkwood/Kconfig | 17 ----- sound/soc/kirkwood/Makefile | 4 -- sound/soc/kirkwood/kirkwood-openrd.c | 109 ----------------------------- sound/soc/kirkwood/kirkwood-t5325.c | 129 ----------------------------------- 4 files changed, 259 deletions(-) delete mode 100644 sound/soc/kirkwood/kirkwood-openrd.c delete mode 100644 sound/soc/kirkwood/kirkwood-t5325.c diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..1f7c7ee3527a 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard <apatard(a)mandriva.com> - * (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/cs42l51.h" - -static int openrd_client_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; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index d213832b0c72..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/alc5623.h" - -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, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -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; -} - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, - .init = t5325_dai_init, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); -- 2.0.0.rc2

3 6

[alsa-devel] [PATCH] ASoC: fix snd_soc_jack_add_gpiods stub
by Arnd Bergmann 27 May '14

27 May '14

f025d3b9c64e1 ("ASoC: jack: Add support for GPIO descriptor defined jack pins") added this new interface, but the stub function provided for non-gpiolib builds was defined in the header file as a global function, leading to lots of "multiple definition of `snd_soc_jack_add_gpiods'" warnings. This adds the obvious "static inline" annotation. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 98dca42..559dc53 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -464,9 +464,9 @@ static inline int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, return 0; } -int snd_soc_jack_add_gpiods(struct device *gpiod_dev, - struct snd_soc_jack *jack, - int count, struct snd_soc_jack_gpio *gpios) +static inline int snd_soc_jack_add_gpiods(struct device *gpiod_dev, + struct snd_soc_jack *jack, + int count, struct snd_soc_jack_gpio *gpios) { return 0; }

2 2