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[alsa-devel] [PATCH] ALSA: hda - Fix loud click noise with IdeaPad 410Y
by Takashi Iwai 07 Mar '14

07 Mar '14

Lenovo IdeaPad 410Y with ALC282 codec makes loud click noises at boot and shutdown. Also, it wrongly misdetects the acpi_thinkpad hook. This patch adds a device-specific fixup for disabling the shutup callback that is the cause of the click noise and also avoiding the thinpad_helper calls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71511 Reported-and-tested-by: Guilherme Amadio <guilherme.amadio(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 850296a1e0ff..8d0a84436674 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3616,6 +3616,19 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } +static void alc_no_shutup(struct hda_codec *codec) +{ +} + +static void alc_fixup_no_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3844,6 +3857,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4020,6 +4034,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_NO_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + }, [ALC269_FIXUP_LENOVO_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4405,6 +4423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.0

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Re: [alsa-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 07 Mar '14

07 Mar '14

Hi Takashi, Thank you once again! Your comments have been very helpful. I now understand how to run ffado-dbus-server/ffado-mixer alongside your ALSA driver. I have pulled your revised Ozonic mixer from ffado-mixer, and when I now invoke ffado-mixer, it properly detects the Ozonic. I can also verify that the mixer control of my audio pass-through (which is plugged into inputs 3 & 4) works as expected. However, I have not yet been able to get audio playback (using "aplay -D plughw:2,0 Desktop/test.wav"). The .wav file is a short, 44.1/16 mono file saved from Audacity, if that's relevant. The terminal output of aplay seems to reflect successful playback, and I hear audible clicks as the file starts and ends its playback. The pass-through is also momentarily muted as aplay starts and stops. I have experimented with every conceivable combination of ffado-mixer settings, ultimately returning to the default settings that it was installed with. When aplay executes, the mixer app shows check marks next to both "Outgoing" and "Incoming". I have been testing the latest git version of the driver (updated this morning), and installing/uninstalling by DKMS. Please, is there anything you can suggest that I should try next? Is there anything obvious that I appear to have overlooked? I will try with a wider range of test materials tomorrow, as well as trying other ALSA apps, but I wanted to get this posted in case that's not the issue. I also recently saw some comments suggesting that pulseaudio be killed. Could that be the issue? I will investigate this tomorrow too. Thank you sincerely, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Sunday, February 23, 2014 9:55 AM To: Brand, Mark (Mr) (Summerstrand Campus South) Cc: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net Subject: Re: Call for testing: ALSA driver for Fireworks/BeBoB based devices Hi Mark, Thanks for your testing. Ozonic is one of devices for which I test. > My testing is probably not particularly rigorous, and I would > sincerely appreciate your advice on what steps to follow > in order to verify your driver's functionality. Try playback/capture PCM samples/MIDI messages with ALSA applications. > In particular, you mention "(to control mixer channels please use > FFADO upstream)". My drivers don't give ways to control device's internal mixer. So you need the other stuff for this purpose. Please refer to this page. You can see some screenshots of ffado-mixer for 'M-Audio Ozonic'. I already commited these improvement to current FFADO trunk except for FW1814/ProjectMiX: MaudioBebob http://subversion.ffado.org/wiki/MaudioBebob > I have not yet set up FFADO on this machine, but I also see under > "Note": "- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' > and 'jackd with Firewire (FFADO) backend'. Both of them try > connecting to the device when another is running." Would you please > explain? How should one rely on the "upstream" mixer functionality > while keeping jackd out of ALSA's way? Devices which my drivers support handle a pair of streams over IEEE1394 bus. PCM samples/MIDI messages are multiplexed into these streams. And snd-fireworks/snd-bebob uses 'duplex streams synchronization'. As a result: - ALSA PCM/MIDI functionality and jackd with firewire backend conflict for its streaming functionality. But: - ALSA PCM/MIDI fuctionality and ffado-dbus-server/ffado-mixer do not interrupt each other. > I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as > ALSA input and output options. However, no audio input is shown when > I try to monitor the audio input I confirm that Audacity has such bugs to detect ALSA PCM character devices. In detail, its PortAudio v19 codes includes bugs. I have no will to investigate this bug because it's not within my work for these drivers. > $aplay -D hw:2,0 test.wav > (where test.wav has been saved from Audacity) > ... and this responds with: > aplay: main:722: audio open error: Input/output error According to aplay codes, snd_pcm_open() generate this error. I guess that another application still uses PCM character device for Ozonic. > $ aplay -D hw:2,0 Desktop/test.wav > Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Mono aplay: set_params:1239: Channels count not available You MUST add 'c' option to indicate the number of channels with 'hw' PCM device. For convinience, use 'plughw' PCM device instead, like: $ aplay -D plughw:2,0 Desktop/test.wav I note an item: - A few days ago, I fixed bug to fail to start stream for Ozonic. See: bebob: fix mappling for MIDI conformant data in stream channels https://github.com/takaswie/snd-firewire-improve/commit/545ab7068f1d53b2b17… Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

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[alsa-devel] Controlling the cards order
by Olivier Langlois 07 Mar '14

07 Mar '14

Hi, I have added a second sound device in my system and it unexpectedly took the first card slot. I have found that it is easy to control the order as long as the devices use different drivers by modifying /etc/modprobe.d/alsa.conf ( http://alsa.opensrc.org/MultipleCards ) but in my situation I have 2 different devices using the same driver. I think that there is hope that I find an answer by digging the more general topic of ordering PCI devices drivers loading but if someone has few pointers to share, any help is welcome! Here are my 2 devices that I would like to change the order: lano1106@whippet2 /proc/asound $ cat cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xfbf20000 irq 84 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfbe60000 irq 88 thank you for your time, Olivier

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[alsa-devel] [PATCH] ASoC: pcm512x: Split out bus drivers
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Move to the new style of defining the bus interfaces in separate modules in order to simplify dependencies. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/pcm512x-i2c.c | 71 ++++++++++++++++++++++ sound/soc/codecs/pcm512x-spi.c | 69 +++++++++++++++++++++ sound/soc/codecs/pcm512x.c | 133 +++-------------------------------------- sound/soc/codecs/pcm512x.h | 9 +++ 6 files changed, 175 insertions(+), 129 deletions(-) create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..3a6046f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,7 +66,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 - select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI + select SND_SOC_PCM512x_I2C if I2C + select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -362,8 +363,19 @@ config SND_SOC_PCM3008 tristate config SND_SOC_PCM512x - tristate "Texas Instruments PCM512x CODECs" - select REGMAP + tristate + +config SND_SOC_PCM512x_I2C + tristate "Texas Instruments PCM512x CODECs - I2C" + depends on I2C + select SND_SOC_PCM512x + select REGMAP_I2C + +config SND_SOC_PCM512x_SPI + tristate "Texas Instruments PCM512x CODECs - SPI" + depends on SPI_MASTER + select SND_SOC_PCM512x + select REGMAP_SPI config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..ac7025c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,6 +54,8 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o 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-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -197,6 +199,8 @@ 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_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-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c new file mode 100644 index 0000000..4d62230 --- /dev/null +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -0,0 +1,71 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "pcm512x.h" + +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_i2c_driver(pcm512x_i2c_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C"); +MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c new file mode 100644 index 0000000..f297058 --- /dev/null +++ b/sound/soc/codecs/pcm512x-spi.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "pcm512x.h" + +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_spi_driver(pcm512x_spi_driver); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index a035ba8..0014d14 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,11 +18,9 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> -#include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> -#include <linux/spi/spi.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> @@ -375,7 +373,7 @@ static const struct regmap_range_cfg pcm512x_range = { .window_start = 0, .window_len = 0x100, }; -static const struct regmap_config pcm512x_regmap = { +const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -391,14 +389,7 @@ static const struct regmap_config pcm512x_regmap = { .cache_type = REGCACHE_RBTREE, }; -static const struct of_device_id pcm512x_of_match[] = { - { .compatible = "ti,pcm5121", }, - { .compatible = "ti,pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(of, pcm512x_of_match); - -static int pcm512x_probe(struct device *dev, struct regmap *regmap) +int pcm512x_probe(struct device *dev, struct regmap *regmap) { struct pcm512x_priv *pcm512x; int i, ret; @@ -510,8 +501,9 @@ err: pcm512x->supplies); return ret; } +EXPORT_SYMBOL_GPL(pcm512x_probe); -static void pcm512x_remove(struct device *dev) +void pcm512x_remove(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -522,6 +514,7 @@ static void pcm512x_remove(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), pcm512x->supplies); } +EXPORT_SYMBOL_GPL(pcm512x_remove); static int pcm512x_suspend(struct device *dev) { @@ -585,122 +578,10 @@ static int pcm512x_resume(struct device *dev) return 0; } -static const struct dev_pm_ops pcm512x_pm_ops = { +const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) }; - -#if IS_ENABLED(CONFIG_I2C) -static int pcm512x_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct regmap *regmap; - - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); - - return pcm512x_probe(&i2c->dev, regmap); -} - -static int pcm512x_i2c_remove(struct i2c_client *i2c) -{ - pcm512x_remove(&i2c->dev); - return 0; -} - -static const struct i2c_device_id pcm512x_i2c_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); - -static struct i2c_driver pcm512x_i2c_driver = { - .probe = pcm512x_i2c_probe, - .remove = pcm512x_i2c_remove, - .id_table = pcm512x_i2c_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -#if defined(CONFIG_SPI_MASTER) -static int pcm512x_spi_probe(struct spi_device *spi) -{ - struct regmap *regmap; - int ret; - - regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); - if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - return ret; - } - - return pcm512x_probe(&spi->dev, regmap); -} - -static int pcm512x_spi_remove(struct spi_device *spi) -{ - pcm512x_remove(&spi->dev); - return 0; -} - -static const struct spi_device_id pcm512x_spi_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { }, -}; -MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); - -static struct spi_driver pcm512x_spi_driver = { - .probe = pcm512x_spi_probe, - .remove = pcm512x_spi_remove, - .id_table = pcm512x_spi_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -static int __init pcm512x_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&pcm512x_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&pcm512x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(pcm512x_modinit); - -static void __exit pcm512x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&pcm512x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&pcm512x_spi_driver); -#endif -} -module_exit(pcm512x_exit); +EXPORT_SYMBOL_GPL(pcm512x_pm_ops); MODULE_DESCRIPTION("ASoC PCM512x codec driver"); MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index ac4a52c..6ee76aa 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,6 +17,9 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X +#include <linux/pm.h> +#include <linux/regmap.h> + #define PCM512x_VIRT_BASE 0x100 #define PCM512x_PAGE_LEN 0x100 #define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) @@ -159,4 +162,10 @@ #define PCM512x_AGBR_SHIFT 0 #define PCM512x_AGBL_SHIFT 4 +extern const struct dev_pm_ops pcm512x_pm_ops; +extern const struct regmap_config pcm512x_regmap; + +int pcm512x_probe(struct device *dev, struct regmap *regmap); +void pcm512x_remove(struct device *dev); + #endif -- 1.9.0

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[alsa-devel] [PATCH] ASoC: pcm512x: Fix duplicate const warning
by Mark Brown 07 Mar '14

07 Mar '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 3a0bbb6..a035ba8 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -203,10 +203,10 @@ static const unsigned int pcm512x_dsp_program_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, - PCM512x_DSP_PROGRAM, 0, 0x1f, - pcm512x_dsp_program_texts, - pcm512x_dsp_program_values); +static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" -- 1.9.0

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[alsa-devel] [PATCH v2 0/2] add support for SPI control mode to tlv320aic23
by Max Filippov 07 Mar '14

07 Mar '14

Hi, this series splits I2C-specific part of tlv320aic23 codec driver into a separate file and adds SPI control mode support. It also updates Kconfigs of existing tlv320aic23 users so that I2C variant is selected. Changes v1->v2: - split i2c interface into separate file; - implement spi interface in a separate file; - update Kconfigs of existing users. Max Filippov (2): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant sound/soc/atmel/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 2 +- sound/soc/codecs/Kconfig | 11 ++++++- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/tlv320aic23-i2c.c | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23-spi.c | 57 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.c | 55 +++++++---------------------------- sound/soc/codecs/tlv320aic23.h | 6 ++++ sound/soc/fsl/Kconfig | 2 +- sound/soc/omap/Kconfig | 4 +-- sound/soc/samsung/Kconfig | 2 +- sound/soc/tegra/Kconfig | 2 +- 12 files changed, 154 insertions(+), 52 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: Check Haswell IPC process_reply/notification return value.
by Liam Girdwood 07 Mar '14

07 Mar '14

Check the return value for error when processing replies and notifications. The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from > Feb 20, 2014, leads to the following imaginary static checker warning: > > sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() > warn: this is always true. > > sound/soc/intel/sst-haswell-ipc.c > 895 /* Handle Immediate reply from DSP Core */ > 896 handled = hsw_process_reply(hsw, ipcx); > ^^^^^^^^^^^^^^^^^ > Returns 1 on success/error and -EIO on error. > > 897 > 898 if (handled) { > 899 /* clear DONE bit - tell DSP we have completed */ > 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, > 901 SST_IPCX_DONE, 0); > 902 > 903 /* unmask Done interrupt */ > 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, > 905 SST_IMRX_DONE, 0); > 906 } > Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1f1576a..f46bb4d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -895,7 +895,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) /* Handle Immediate reply from DSP Core */ handled = hsw_process_reply(hsw, ipcx); - if (handled) { + if (handled > 0) { /* clear DONE bit - tell DSP we have completed */ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, SST_IPCX_DONE, 0); @@ -913,7 +913,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) handled = hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled) { + if (handled > 0) { sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); -- 1.8.3.2

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Correct type specifier in printf
by Charles Keepax 07 Mar '14

07 Mar '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 937af6f..bb5f7b4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -706,7 +706,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) to_write); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", file, regions, to_write, offset, region_name, ret); -- 1.7.2.5

2 1

Re: [alsa-devel] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Ben Hutchings 06 Mar '14

06 Mar '14

I'm looking at commit 34354792432b6e0a3b156819a1a19716c50d3473 and include/linux/mfd/arizona/registers.h, and wondering what exactly this the register patch is changing. It looks like there are some registers not named elsewhere (addresses 0x0049, 0x004a, 0x0057, 0x0058). What about the higher addresses? Are these more registers, or is this patching firmware in device memory? Ben. -- Ben Hutchings The world is coming to an end. Please log off.

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic23: add support for SPI control mode
by Max Filippov 06 Mar '14

06 Mar '14

tlv320aic23 chip control interface may work in either I2C or SPI mode depending on the MODE pin state. Functionality and register layout are independent of the control mode. Provide i2c and spi driver entry points when CONFIG_I2C and CONFIG_SPI_MASTER are enabled respectively. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic23.c | 58 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 56 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..6d82de5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C - select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC23 if SND_SOC_I2C_AND_SPI select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc..2418260 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -26,6 +26,7 @@ #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -617,12 +618,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; +#ifdef CONFIG_I2C /* * If the i2c layer weren't so broken, we could pass this kind of data * around */ -static int tlv320aic23_codec_probe(struct i2c_client *i2c, - const struct i2c_device_id *i2c_id) +static int tlv320aic23_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) { struct aic23 *aic23; int ret; @@ -661,12 +663,62 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .driver = { .name = "tlv320aic23-codec", }, - .probe = tlv320aic23_codec_probe, + .probe = tlv320aic23_i2c_probe, .remove = __exit_p(tlv320aic23_i2c_remove), .id_table = tlv320aic23_id, }; module_i2c_driver(tlv320aic23_i2c_driver); +#endif /* CONFIG_I2C */ + +#ifdef CONFIG_SPI_MASTER +static int aic23_spi_probe(struct spi_device *spi) +{ + struct aic23 *aic23; + int ret; + + dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); + + spi->bits_per_word = 16; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + /* Allocate driver data */ + aic23 = devm_kzalloc(&spi->dev, sizeof(struct aic23), GFP_KERNEL); + if (!aic23) + return -ENOMEM; + + aic23->regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + + /* Initialize the driver data */ + dev_set_drvdata(&spi->dev, aic23); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); + return ret; +} + +static int aic23_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver aic23_spi = { + .driver = { + .name = "tlv320aic23", + .owner = THIS_MODULE, + }, + .probe = aic23_spi_probe, + .remove = aic23_spi_remove, +}; + +module_spi_driver(aic23_spi); +#endif /* CONFIG_SPI_MASTER */ MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

3 5

[alsa-devel] [PATCH 1/7] ASoC: Handle ignore_pmdown_time for CODEC to CODEC links
by Lars-Peter Clausen 06 Mar '14

06 Mar '14

For CODEC to CODEC links we should only immediately power down if both CODECs are configured to ignore the power down delay. Factor the logic for this into a helper function that can be used for both compressed and normal PCMs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 ++ sound/soc/soc-compress.c | 3 +-- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++++++-- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..93c31c7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -413,6 +413,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link); +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5e9690c..ef585af 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -235,8 +235,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..f098c80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,30 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay + * @rtd: The ASoC PCM runtime that should be checked. + * + * This function checks whether the power down delay should be ignored for a + * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has + * been configured to ignore the delay, or if none of the components benefits + * from having the delay. + */ +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) +{ + bool ignore = true; + + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) + return true; + + if (rtd->cpu_dai->codec) + ignore &= rtd->cpu_dai->codec->ignore_pmdown_time; + + ignore &= rtd->codec_dai->codec->ignore_pmdown_time; + + return ignore; +} + +/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -496,8 +520,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, -- 1.8.0

2 7

[alsa-devel] [PATCH] ASoC: dapm: Sprinkle lockdep asserts through the code
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> Try to spot locking issues by asserting that the DAPM mutex is held when it should be. There's a bit of fun due to us not requiring the lock to be held prior to the card being instantiated which mean we need to wrap the assert in some paths and this isn't methodical by any stretch of the imagination. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- I've only compile tested this so it's entirely possible there's some oversights here - at this point I'm mostly just throwing it out there for people to play with as a work in progress. sound/soc/soc-dapm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3728e21..c8a780d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -111,6 +111,12 @@ static int dapm_down_seq[] = { [snd_soc_dapm_post] = 14, }; +static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) +{ + if (dapm->card && dapm->card->instantiated) + lockdep_assert_held(&dapm->card->dapm_mutex); +} + static void pop_wait(u32 pop_time) { if (pop_time) @@ -144,6 +150,8 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { + dapm_assert_locked(w->dapm); + if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", w->name, reason); @@ -356,6 +364,8 @@ static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; + lockdep_assert_held(&card->dapm_mutex); + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { @@ -1750,6 +1760,8 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; + lockdep_assert_held(&card->dapm_mutex); + trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { @@ -2045,6 +2057,8 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) @@ -2095,6 +2109,8 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; @@ -2260,6 +2276,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + dapm_assert_locked(dapm); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Staticise dapm_mark_dirty()
by Mark Brown 06 Mar '14

06 Mar '14

From: Mark Brown <broonie(a)linaro.org> The function is not called outside soc-dapm.c so there is no need for it to be exported. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 3 +-- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 04d32d8..ef78f56 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -449,7 +449,6 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8ba30..3728e21 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -142,7 +142,7 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -150,7 +150,6 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } -EXPORT_SYMBOL_GPL(dapm_mark_dirty); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) { -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER
by Takashi Iwai 06 Mar '14

06 Mar '14

cs4271_common_probe() is called from cs4271_i2c_probe() but defined in CONFIG_SPI_MASTER block, thus it results in a build error when CONFIG_SPI_MASTER=n: sound/soc/codecs/cs4271.c:721:2: error: implicit declaration of function ‘cs4271_common_probe’ [-Werror=implicit-function-declaration] Move the function out of #if block. Fixes: d6cf89ee07cb ('ASoC: cs4271: claim reset GPIO in bus probe function') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/cs4271.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 96c309777208..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -612,22 +612,6 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, - .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { @@ -658,6 +642,22 @@ static int cs4271_common_probe(struct device *dev, return 0; } +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; -- 1.9.0

4 3

[alsa-devel] [patch] ASoC: Baytrail: fix error handling in sst_byt_dsp_init()
by Dan Carpenter 06 Mar '14

06 Mar '14

Calling "kfree(byt)" is a double free because that was allocated with devm_kzalloc(). There were a couple places which leak "byt->msg". That memory is allocated in msg_empty_list_init(). Fixes: f7d01fd6754c ('ASoC: Intel: Add Intel Baytrail SST DSP IPC support') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index c12e194bbc6b..d0eaeee21be4 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -796,7 +796,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) err = msg_empty_list_init(byt); if (err < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&byt->kworker); @@ -806,7 +806,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (IS_ERR(byt->tx_thread)) { err = PTR_ERR(byt->tx_thread); dev_err(byt->dev, "error failed to create message TX task\n"); - goto list_err; + goto err_free_msg; } init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); @@ -816,7 +816,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto list_err; + goto err_free_msg; } /* keep the DSP in reset state for base FW loading */ @@ -848,9 +848,9 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +err_free_msg: kfree(byt->msg); -list_err: - kfree(byt); + return err; } EXPORT_SYMBOL_GPL(sst_byt_dsp_init);

3 2

[alsa-devel] UCM: How does its api relate to ALSA lib api?
by kaweka 05 Mar '14

05 Mar '14

Hi, I am a newcomer if it goes to following topics: ALSA, audio in Linux, audio in embedded Linux. However several years of experience in audio development on other platforms. As next a development of audio applications in embedded Linux is the task. However have few gaps in understanding of the audio system architecture. It would be helpful to have answers to following questions. For any hints lot of thanks in advance! * There is well-known handshaking application-ALSA lib based on amonst others snd_pcm_open snd_pcm_hw_params_alloca snd_pcm_hw_params_any snd_pcm_hw_params_set_... snd_pcm_hw_params snd_pcm_hw_params_get_... snd_pcm_hw_params_is_... snd_pcm_hw_params_can_... snd_pcm_close * Does the UCM and its api complement the api and handshaking as above. Or is it rather a question of replacement? * If neither nor, how to describe best the relation between UCM api and ALSA lib api? * How does the entire handshaking application-audio system look like if to use ALSA and UCM api? * An use case seems to be just a container for verb, device, modifier. Is an use case introducing any functionality to be found neither in verb, nor in device, nor in modifier? * For what reason does the term / definition of use case exists at all. Will verb, device, modifier not be enough? * What is the difference between switching to other use case and modifying a use case? Why not to switch to other use case in situations where modifier is used? * QoS api seems to introduce some redundancy. If voice, or music, or other - this information can be gathered from use case. A use case does already read voice, music, other! Why is qos api necessary? Best Regards kaweka

1 0

[alsa-devel] Add "cval->res = 384" quirk for Logitech Webcam C500
by Simon 05 Mar '14

05 Mar '14

Hello, Since some time I am successfully using this workaround http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037657.html to solve the problem for the Logitech C500 Webcam (046d:0807). Would it be possible to add this device to the list included in the mixer.c file? Thanks

2 1

[alsa-devel] [patch] ALSA: echoaudio: use after free on error
by Dan Carpenter 05 Mar '14

05 Mar '14

There are some places where we dereference "chip" in the error message but we've already freed it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 166ec0c120d4..9f10c9e0df5e 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1990,8 +1990,8 @@ static int snd_echo_create(struct snd_card *card, if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz, ECHOCARD_NAME)) == NULL) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot get memory region\n"); + snd_echo_free(chip); return -EBUSY; } chip->dsp_registers = (volatile u32 __iomem *) @@ -1999,8 +1999,8 @@ static int snd_echo_create(struct snd_card *card, if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; @@ -2012,8 +2012,8 @@ static int snd_echo_create(struct snd_card *card, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), sizeof(struct comm_page), &chip->commpage_dma_buf) < 0) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot allocate the comm page\n"); + snd_echo_free(chip); return -ENOMEM; } chip->comm_page_phys = chip->commpage_dma_buf.addr; @@ -2291,8 +2291,8 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_echo_free(chip); dev_err(chip->card->dev, "cannot grab irq\n"); + snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq;

3 3

Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 05 Mar '14

05 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Could you submit v2 patches to ML? The patches need review, and I want get them acked by Clemens before applying. thanks, Takashi > > ---------------------------------------------------------------- > The following changes since commit 31278997add61ee13b709e274934c7b0085accce: > > ALSA: hda/realtek - Add headset quirk for Dell DT (2014-03-03 10:48:55 > +0100) > > are available in the git repository at: > > https://github.com/takaswie/sound.git request > > for you to fetch changes up to 5b354e20ec3c64fa0a8fa7fbc7355201d881a8cc: > > bebob: Send a cue to load firmware for M-Audio Firewire series > (2014-03-05 12:00:32 +0900) > > ---------------------------------------------------------------- > Takashi Sakamoto (39): > firewire-lib: Rename functions, structure, member for AMDTP > 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 sort function for transmitted packet > firewire-lib: Add transfer delay to synchronized duplex streams > firewire-lib: Add support for channel mapping > 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 some AV/C general commands > firewire-lib: Add quirks for Fireworks > firewire-lib: Add a fallback at RCODE_CANCELLED > fireworks: Add skelton for Fireworks based devices > fireworks: Add transaction and some commands > fireworks: Add connection and stream management > fireworks: Add proc interface for debugging purpose > fireworks: Add MIDI interface > 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: 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: Send a cue to load firmware for M-Audio Firewire series > > include/uapi/sound/asound.h | 4 +- > include/uapi/sound/firewire.h | 22 +- > sound/firewire/Kconfig | 59 ++ > sound/firewire/Makefile | 2 + > sound/firewire/amdtp.c | 891 > +++++++++++++++++------ > sound/firewire/amdtp.h | 175 ++++- > sound/firewire/bebob/Makefile | 4 + > sound/firewire/bebob/bebob.c | 450 ++++++++++++ > sound/firewire/bebob/bebob.h | 260 +++++++ > sound/firewire/bebob/bebob_command.c | 354 +++++++++ > sound/firewire/bebob/bebob_focusrite.c | 287 ++++++++ > sound/firewire/bebob/bebob_hwdep.c | 197 +++++ > sound/firewire/bebob/bebob_maudio.c | 828 > +++++++++++++++++++++ > sound/firewire/bebob/bebob_midi.c | 170 +++++ > sound/firewire/bebob/bebob_pcm.c | 451 ++++++++++++ > sound/firewire/bebob/bebob_proc.c | 201 +++++ > sound/firewire/bebob/bebob_stream.c | 869 > ++++++++++++++++++++++ > sound/firewire/bebob/bebob_terratec.c | 68 ++ > sound/firewire/bebob/bebob_yamaha.c | 50 ++ > sound/firewire/cmp.c | 229 ++++-- > sound/firewire/cmp.h | 14 +- > sound/firewire/dice.c | 46 +- > sound/firewire/fcp.c | 160 +++- > sound/firewire/fcp.h | 21 + > sound/firewire/fireworks/Makefile | 4 + > sound/firewire/fireworks/fireworks.c | 326 +++++++++ > sound/firewire/fireworks/fireworks.h | 223 ++++++ > 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 | 337 +++++++++ > sound/firewire/fireworks/fireworks_transaction.c | 327 +++++++++ > sound/firewire/lib.c | 6 +- > sound/firewire/lib.h | 1 + > sound/firewire/speakers.c | 93 +-- > 37 files changed, 8307 insertions(+), 402 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 >

2 3

[alsa-devel] [PATCH] ALSA: emu10k1: Fix possible NULL dereference
by Takashi Iwai 05 Mar '14

05 Mar '14

The previous dev_err() conversion resulted in a code that may give NULL dereference in snd_emu10k1_ptr_write(). Since it's a sanity check, better to be replaced with a debug macro like other places in this driver. Fixes: 6f002b02166c ('ALSA: emu10k1: Use standard printk helpers') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/io.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 81c75b4528d9..706b4f0c6806 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -71,11 +71,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i unsigned long flags; unsigned int mask; - if (!emu) { - dev_err(emu->card->dev, "ptr_write: emu is null!\n"); - dump_stack(); + if (snd_BUG_ON(!emu)) return; - } mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK; regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK); -- 1.9.0

1 0

[alsa-devel] [patch] ALSA: lola: NULL deref on allocation error
by Dan Carpenter 05 Mar '14

05 Mar '14

"chip" is NULL here. We don't need a printk here because kmalloc() has it built in. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index d63434d8c3fb..68824cdd137d 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -586,7 +586,6 @@ static int lola_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - dev_err(chip->card->dev, "cannot allocate chip\n"); pci_disable_device(pci); return -ENOMEM; }

2 1

Re: [alsa-devel] ALSA: emu10k1: Use standard printk helpers
by Dan Carpenter 05 Mar '14

05 Mar '14

Hello Takashi Iwai, The patch 6f002b02166c: "ALSA: emu10k1: Use standard printk helpers" from Feb 25, 2014, leads to the following static checker warning: sound/pci/emu10k1/io.c:75 snd_emu10k1_ptr_write() error: we previously assumed 'emu' could be null (see line 74) sound/pci/emu10k1/io.c 68 void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data) 69 { 70 unsigned int regptr; 71 unsigned long flags; 72 unsigned int mask; 73 74 if (!emu) { ^^^^ 75 dev_err(emu->card->dev, "ptr_write: emu is null!\n"); ^^^^^^^^^^^^^^ Dereference. 76 dump_stack(); 77 return; 78 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Add missing loopback merge path for AD1884/1984 codecs
by Takashi Iwai 05 Mar '14

05 Mar '14

The mixer widget (NID 0x20) of AD1884 and AD1984 codecs isn't connected directly to the actual I/O paths but only via another mixer widget (NID 0x21). We need a similar fix as we did for AD1882. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index df3652ad15ef..498971095869 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1092,6 +1092,7 @@ static int patch_ad1884(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; + spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); -- 1.9.0

1 1

[alsa-devel] Xonar DG backports
by Clemens Ladisch 05 Mar '14

05 Mar '14

Hi Roman, I backporting the bugfixes in your patches to the stable kernels. However, I'm unable to see what was changed to enable the headphone outputs. What particular thing did the old driver wrong? Regards, Clemens

2 1

[alsa-devel] Audio-Bug
by Silver Death 05 Mar '14

05 Mar '14

Hi there, many people have this bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 I would fix it, but I have no knowledge in C and about the kernel internal. I also didn't found any documentation which I could use to translate the datasheet into the source file. So I gentle ask you, if you could as a maintainer fix this bug or at least give me some information/instruction/documentation. Thanks in advance. With best regards, Simon

1 0

Re: [alsa-devel] Question on compress offload framework memory corruption
by Banajit Goswami 05 Mar '14

05 Mar '14

Hi Liam, > On Thu, Dec 19, 2013 at 10:55:08AM +0000, Mark Brown wrote: > > On Wed, Dec 18, 2013 at 01:17:51PM -0000, gsantosh(a)codeaurora.org wrote: > > > Hi, > > > > > > I have following questions in the compressed offload framework API. > > > > > > static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, > > > struct snd_compr_params *params) { ... > > > hw_params = kzalloc(sizeof(*hw_params), GFP_KERNEL); > > > if (hw_params == NULL) > > > return -ENOMEM; > > > /*1st question is, what is the use of above allocated memory I do > > > not see this being used in this function*/ > > > > The code you're quoting there isn't in mainline. I've also added > > Vinod to the CCs, you probably want to include him in any discussion > > of the compressed API. > > That's right, this is from DPCM compressed updated done by Liam, so he is > the best person to comment on this bit, and somehow his address in CC has > gone bad, i have added him in to list > But yes looking at the source, i agree that two are interlinked. > I guess we need to poupulate the hw_params and then use that to copy, that > would be the right thing to do here and yes this is not correct and not > sure how this is working... Continuing the discussion on the "memcpy" function in Santosh's 2nd question- > > > memcpy(&fe->dpcm[fe_substream->stream].hw_params, params, > > > sizeof(struct snd_pcm_hw_params)); > > > > > > /* 2nd question is > > > in above memcpy there is parameter mismatch > > > &fe->dpcm[fe_substream->stream].hw_params is of structure type struct > > > snd_pcm_hw_params > > > > > > params argument is of the structure type struct snd_compr_params > > > the definition of the two structures are different, how this is > > > working? > > > this will over right the destination with junk value which will return > > > in error when this memory is accessed, after memcpy this cpu_dai > > > hw_params > > > returing with EINVAL error, please explain why this is done this way. > > > */ I can see that a later version of the patch adds the function soc_compr_set_params_fe() at- http://permalink.gmane.org/gmane.linux.alsa.devel/117716 with the below code replaces the "memcpy" function with a "memset"- + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); It's evident that the code relies on machine driver fixup function to populate the parameters of hw_params before propagating those to back-end. I think rather than that, like Vinod also mentioned, the hw_params should have been populated with individual parameters which are relevant. Let me know your comment on this. Thanks, Banajit The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] Update patch for codecs alias name for Dell
by Kailang 05 Mar '14

05 Mar '14

Hi Takashi, Dell assigned more codecs for alias name. I update in attach patch. Thanks. Kailang

3 7

[alsa-devel] [PATCH v5 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../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/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 534 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + 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/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2691 insertions(+), 1 deletions(-) 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/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt 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/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 create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

2 8

[alsa-devel] [PATCH 00/15] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 05 Mar '14

05 Mar '14

Hi, Here is a series of cleanups for the fsl-ssi driver. It would be great if someone with PPC hardware and this fsl-ssi unit could please test this series. I am not sure if the regmap patch works as expected on PPC as well. I tested this series on imx53 quick starter board and a custom board. Both with SSI as I2S slave. Regards, Markus Markus Pargmann (15): ASoC: fsl-ssi: Use regmap ASoC: fsl-ssi: Remove fsl_ssi_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 .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 1167 ++++++++------------ sound/soc/fsl/fsl_ssi.h | 112 +- sound/soc/fsl/fsl_ssi_dbg.c | 163 +++ 6 files changed, 750 insertions(+), 716 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.8.5.3

4 21

[alsa-devel] [PATCH 00/10] Simplify the code of TDM slot setting
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (10): ASoC: core: Add snd_soc_dai_set_tdm_slot_xlate(). ASoC: fsl-utils: Add fsl_asoc_of_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .of_xlate_tdm_slot_mask() support. ASoC: fsl-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .of_xlate_tdm_slot_mask() support. ASoC: simple-card: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad1836: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: blackfin: bf5xx-ad193x: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: eukrea-tlv320: Use snd_soc_dai_set_tdm_slot_xlate() ASoC: fsl: wm1133-ev1: Use snd_soc_dai_set_tdm_slot_xlate() include/sound/soc-dai.h | 3 +++ sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- sound/soc/fsl/Kconfig | 3 +++ sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 ++ sound/soc/fsl/fsl_ssi.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/fsl/wm1133-ev1.c | 11 +---------- sound/soc/generic/simple-card.c | 7 +++---- sound/soc/soc-core.c | 33 ++++++++++++++++++++++++++++----- 13 files changed, 78 insertions(+), 24 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH] ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT
by Kuninori Morimoto 05 Mar '14

05 Mar '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 30d0341e7da0c012f64fb290dd1153360fb49a8d (ASoC: simple-card: simplify the daifmt code) simplify cpu/codec dai_fmt which consists from dai specific format + common format. But, it didn't care about non-DT case. This patch fixes it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 034a2b7..7cabcc5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -307,6 +307,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) sizeof(priv->cpu_dai)); memcpy(&priv->codec_dai, &cinfo->codec_dai, sizeof(priv->codec_dai)); + + priv->cpu_dai.fmt |= cinfo->daifmt; + priv->codec_dai.fmt |= cinfo->daifmt; } /* -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 05 Mar '14

05 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v4: 1. Use the generic dmaengine pcm codes instead of sirf pcm driver. 2. Add TLV information into internal audio codec driver. 3. Use the MMIO regmap instead of custom read and write callbacks. 4. Remove spin_lock. 5. Change some variable names. 6. The SiRF internal audio driver split into codec, port. 7. Add binding documents. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../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/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 585 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + 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/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 438 +++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 +++ sound/soc/sirf/sirf-usp.c | 448 +++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 ++++++++++ 24 files changed, 2745 insertions(+), 0 deletions(-) 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/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt 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/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 create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

4 18

[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 05 Mar '14

05 Mar '14

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. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ 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); @@ -2018,6 +2019,7 @@ 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); @@ -2082,6 +2084,7 @@ 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"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] [RFC][PATCH 0/3] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (3): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: codec: remove the set_cache_io() entirely from CODEC ASoC probe. ASoC: core: remove the default regmap I/O setting up from soc_probe_codec() include/sound/soc.h | 3 +++ sound/soc/codecs/cq93vc.c | 6 ++++-- sound/soc/codecs/mc13783.c | 10 ++-------- sound/soc/codecs/wm5102.c | 6 ++---- sound/soc/codecs/wm5110.c | 6 ++---- sound/soc/codecs/wm8350.c | 6 ++++-- sound/soc/codecs/wm8400.c | 6 ++++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8997.c | 7 ++----- sound/soc/soc-core.c | 21 +++++++++++---------- 10 files changed, 36 insertions(+), 39 deletions(-) -- 1.8.4

3 5

[alsa-devel] [PATCH] ASoC: ak4104: Remove superfluous codec->control_data initialization
by Lars-Peter Clausen 05 Mar '14

05 Mar '14

The driver uses automatic IO setup, which will also initialize the control_data field of the CODEC, no need to do it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4104.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b4819dc..10adf25 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ak4104->regmap; - /* set power-up and non-reset bits */ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: io: Remove hw_read() operation
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> We now no longer have any users of hw_read() in the kernel so remove the code in order to prevent any new users being added. Users should be using regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 - sound/soc/soc-io.c | 13 ------------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 56c4c71..dc36331 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,7 +706,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ hw_write_t hw_write; - unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c3b6a0a..3c1de9c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -26,18 +26,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, return regmap_write(codec->control_data, reg, value); } -static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -64,7 +52,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int ret; codec->write = hw_write; - codec->read = hw_read; switch (control) { case SND_SOC_REGMAP: -- 1.9.0

2 6

[alsa-devel] [PATCH] ASoC: fsl: Don't select FIQ from Eukrea
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> There is no point in using FIQ if DMA is available (it is selected) and selecting FIQ currently breaks the build on non-i.MX platforms. If FIQ is actually required the build will need to be restricted or have a select for the relevant FIQ code adding. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a7d6ab6..b244830 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_EUKREA_TLV320 || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 - select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI select SND_SOC_FSL_SSI -- 1.9.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 04 Mar '14

04 Mar '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index add088b..f505b1f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -283,13 +283,13 @@ static int rsnd_gen2_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start, + dev_dbg(dev, "SRU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; -- Regards, Laurent Pinchart

1 0

[alsa-devel] SNDRV_PCM_INFO_BATCH
by David Henningsson 04 Mar '14

04 Mar '14

So, PulseAudio removed timer scheduling for drivers with SNDRV_PCM_INFO_BATCH, and released 5.0. Meanwhile, USB drivers still have SNDRV_PCM_INFO_BATCH, so this leads to performance regressions for USB sound cards. The question remains - what does SNDRV_PCM_INFO_BATCH really mean? In the documentation, we have this: "int snd_pcm_hw_params_is_batch (const snd_pcm_hw_params_t *params) Check, if hardware does double buffering for data transfers for given configuration." What PulseAudio was looking for was more something like "what accuracy does the pointer callback give us". When we asked this question at the latest audio meeting, SNDRV_PCM_INFO_BATCH was given as the answer, i e, if this flag is set, pointer callback have no more accuracy than period boundaries. At least that's how I understood it. So, either the documentation for snd_pcm_hw_params_is_batch is correct, in which case PulseAudio should revert the tsched patch. Or the answer given at the audio meeting was correct, in which case we need to fix the USB driver to stop sending the SNDRV_PCM_INFO_BATCH flag. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 2

[alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire devices
by Takashi Sakamoto 04 Mar '14

04 Mar '14

This is a pull-request of my 39 patches to enhance support for Firewire devices. These patches extend firewire-lib, add new drivers (fireworks/bebob). These drivers can handle playback/capture of PCM samples/MIDI messages and can support 70-80 models totally. These drivers are designed not to interrupt FFADO (user-land driver project) functionality except for playbacking/capturing. These drivers basically give no way to control device's internal mixer. For this purpose, users need to use applications such like ffado-dbus-server/ffado-mixer, or develop software to execute AV/C commands and device specific commands. Updates since RFC v3: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… firewire-lib: - improve operation for oPCR - add fallbacks at transaction timeout for device's quirk fireworks: - add counter to maintain streams - use memdup_user() for hwdep interface bebob: - add counter to maintain streams - fix failure to handle bus reset ---------------------------------------------------------------- The following changes since commit dde9c38779c1f30253485a160facc211b20ca222: Merge branch 'for-linus' (2014-02-27 12:46:40 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to 0ed07c65fef2f80fcfc6223b0d689fca56fa6323: bebob: Add support for M-Audio special Firewire series (2014-02-28 11:30:52 +0900) ---------------------------------------------------------------- Takashi Sakamoto (39): firewire-lib: Rename functions, structure, member for AMDTP 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 sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping 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 some AV/C general commands firewire-lib: Add quirks for Fireworks firewire-lib: Add a fallback at RCODE_CANCELLED fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface 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: 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: Send a cue to load firmware for M-Audio Firewire series bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 891 ++++++++++++++++------ sound/firewire/amdtp.h | 175 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 450 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 170 +++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 179 +++++ sound/firewire/bebob/bebob_stream.c | 869 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 229 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 160 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 324 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ 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 | 208 +++++ sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/lib.c | 6 +- sound/firewire/lib.h | 1 + sound/firewire/speakers.c | 93 +-- 37 files changed, 8356 insertions(+), 402 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

4 67

[alsa-devel] [PATCH RFC 0/5] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 04 Mar '14

04 Mar '14

This patch set adds audio support for AM43xx-ePOS-EVM. I'll mail the dts and defconfig changes once these have been merged. Jyri Sarha (5): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: tlv320aic31xx: Add codec driver to Makefile and Kconfig ASoC: tlv320aic31xx: Add DT binding document ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1360 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 265 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 9 files changed, 1748 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

3 10

[alsa-devel] [PATCH] Wrap hw_ptr to boundary in pcm_ioplug
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

The function snd_pcm_ioplug_hw_ptr_update() always increased the hw_ptr by delta, without wrapping it to the boundary. This would lead to problems when after many hours, the hw_ptr would overflow. --- src/pcm/pcm_ioplug.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a90c844..c1c3a98 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -60,7 +60,7 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) delta = hw - io->last_hw; else delta = pcm->buffer_size + hw - io->last_hw; - io->data->hw_ptr += delta; + snd_pcm_mmap_hw_forward(io->data->pcm, delta); io->last_hw = hw; } else io->data->state = SNDRV_PCM_STATE_XRUN; -- 1.7.0.4 -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

2 1

[alsa-devel] Fix for busy loop at boundary wrapping with ioplug based plugins
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

Hello everybody, I have found an odd problem during long testing of our device, that suddenly starts misbehaving after several hours of use. What happened is: A client opens a device based on the jack plugin for reading, and keeps reading from it. After 10 hours 55 minutes, the snd_pcm_readi() functions starts to report data to be always available for read, and makes the reader use all the cpu it can. My patch will follow shortly, please tell me what you think of it. Best regards, Luciano -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

1 0

[alsa-devel] [PATCH 2/4] ARM: dove: prepare new Dove DT Kconfig variable
by Sebastian Hesselbarth 04 Mar '14

04 Mar '14

DT-enabled Dove will move over from ARCH_DOVE in mach-dove to MACH_DOVE in mach-mvebu. As non-DT ARCH_DOVE will stay to rot for a while, add a new DT-only MACH_DOVE to mmc, thermal, wdt, and asoc Kconfigs. Signed-off-by: Sebastian Hesselbarth <sebastian.hesselbarth(a)gmail.com> --- Cc: Chris Ball <chris(a)printf.net> Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: Eduardo Valentin <eduardo.valentin(a)ti.com> Cc: Wim Van Sebroeck <wim(a)iguana.be> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Gregory Clement <gregory.clement(a)free-electrons.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-watchdog(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org --- drivers/mmc/host/Kconfig | 2 +- drivers/thermal/Kconfig | 2 +- drivers/watchdog/Kconfig | 2 +- sound/soc/kirkwood/Kconfig | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/mmc/host/Kconfig b/drivers/mmc/host/Kconfig index 1384f67abe21..a69e48b582e5 100644 --- a/drivers/mmc/host/Kconfig +++ b/drivers/mmc/host/Kconfig @@ -168,7 +168,7 @@ config MMC_SDHCI_ESDHC_IMX config MMC_SDHCI_DOVE tristate "SDHCI support on Marvell's Dove SoC" - depends on ARCH_DOVE + depends on ARCH_DOVE || MACH_DOVE depends on MMC_SDHCI_PLTFM select MMC_SDHCI_IO_ACCESSORS help diff --git a/drivers/thermal/Kconfig b/drivers/thermal/Kconfig index 35c066489a19..07b5e56e9a97 100644 --- a/drivers/thermal/Kconfig +++ b/drivers/thermal/Kconfig @@ -150,7 +150,7 @@ config KIRKWOOD_THERMAL config DOVE_THERMAL tristate "Temperature sensor on Marvell Dove SoCs" - depends on ARCH_DOVE + depends on ARCH_DOVE || MACH_DOVE depends on OF help Support for the Dove thermal sensor driver in the Linux thermal diff --git a/drivers/watchdog/Kconfig b/drivers/watchdog/Kconfig index 79d25894343a..5659c62a175c 100644 --- a/drivers/watchdog/Kconfig +++ b/drivers/watchdog/Kconfig @@ -292,7 +292,7 @@ config DAVINCI_WATCHDOG config ORION_WATCHDOG tristate "Orion watchdog" - depends on ARCH_ORION5X || ARCH_KIRKWOOD || ARCH_DOVE + depends on ARCH_ORION5X || ARCH_KIRKWOOD || ARCH_DOVE || MACH_DOVE select WATCHDOG_CORE help Say Y here if to include support for the watchdog timer diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 78ed4a42ad21..a1506d57e646 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST + depends on ARCH_KIRKWOOD || ARCH_DOVE || MACH_DOVE || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- 1.8.5.3

4 9

[alsa-devel] [PATCH] ASoC: da732x: Replace hw_read usage with snd_soc_read()
by Mark Brown 04 Mar '14

04 Mar '14

From: Mark Brown <broonie(a)linaro.org> Pre-merge code was using direct hw_read() calls as an out of framework way of doing volatile register I/O when not using regmap. This has never functioned correctly in mainline due to the regmap conversion, the hw_read() implementation still does caching. In order to facilitate removal of the subsystem level I/O code convert to use snd_soc_read(), there should be no functional impact. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/da732x.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 4d1c302..14acb3a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1297,9 +1297,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check DAC offset sign */ - sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + sign[DA732X_HPL_DAC] = snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); - sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + sign[DA732X_HPR_DAC] = snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); /* Binary search DAC offset values (both channels at once) */ @@ -1316,10 +1316,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) offset[DA732X_HPL_DAC] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) offset[DA732X_HPR_DAC] &= ~step; @@ -1360,9 +1360,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check output offset sign */ - sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO; - sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO; snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | @@ -1383,10 +1383,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL) & + if ((snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) offset[DA732X_HPL_AMP] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR) & + if ((snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) offset[DA732X_HPR_AMP] &= ~step; -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: hda - add automute fix for another dell AIO model
by Hui Wang 04 Mar '14

04 Mar '14

When plugging a headphone or headset, lots of noise is heard from internal speaker, after changing the automute via amp instead of pinctl, the noise disappears. BugLink: https://bugs.launchpad.net/bugs/1268468 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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2056b7e..b501141 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5263,7 +5263,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), - SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2] ASoC: snappercl15: Convert to table based DAPM setup
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Tab align the right side of the field initializers --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7..5b68b10 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int snappercl15_tlv320aic23_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_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", @@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-i2s", - .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, @@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = { .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int snappercl15_probe(struct platform_device *pdev) -- 1.8.0

1 0

Re: [alsa-devel] change kmalloc into vmalloc for large memory allocations
by 'gregkh＠linuxfoundation.org' 04 Mar '14

04 Mar '14

On Mon, Mar 03, 2014 at 10:51:23AM +0800, Wang, Yalin wrote: > Hi greg, > > I am sorry, > I make a mistake , > You are right , > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/us… > > this one should not changed to use vmalloc, > the buffer will be used by DMA， Which is why a wrapper function will never work. > others should be safe to change: > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/sound/soc/… > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… > > https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/in… Then send individual patches for these and see what happens. greg k-h

2 3

[alsa-devel] [PATCH] ASoC: cache: Do the codec->reg_cache zero pionter check
by Xiubo Li 04 Mar '14

04 Mar '14

For the snd_soc_cache_init(), the reg_size maybe zero and then the value of codec->reg_cache, which is alloced via kzalloc, maybe equal to ZERO_SIZE_PTR. If the reg parameter of snd_soc_cache_write() is large enough, the cache[idx] = val maybe cause the kernel crash... So this patch fix this via doing the zero pionter check of it. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-cache.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 375dc6d..bfed3e4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -96,8 +96,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) { dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", codec->name); - if (!codec->reg_cache) - return 0; + kfree(codec->reg_cache); codec->reg_cache = NULL; return 0; @@ -117,8 +116,9 @@ int snd_soc_cache_read(struct snd_soc_codec *codec, return -EINVAL; mutex_lock(&codec->cache_rw_mutex); - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; @@ -136,8 +136,9 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { mutex_lock(&codec->cache_rw_mutex); - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; -- 1.8.4

3 2

[alsa-devel] [PATCH RESEND] sound: omap: n810: fix init with DT boot
by Aaro Koskinen 04 Mar '14

04 Mar '14

Since 3.14-rc1 only DT boot has been supported on N810, so this file fails to init. Make a minimal fix to retain functionality. This file should be properly converted to DT in longer term. There seems to be still other unresolved issues with N810 audio support, but this patch is needed at minimum as otherwise the machine driver probing would completely fail. Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> Acked-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Previous discussion: http://marc.info/?l=linux-omap&m=139387208426122&w=2 sound/soc/omap/n810.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e4..d163e18 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm5192, wm5110 and wm8997 drivers for this by using arizona->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffc..3410905 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -582,7 +582,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf1..d7bf884 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -136,7 +136,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442c..e10f44d 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -86,7 +86,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1'
by Lars-Peter Clausen 04 Mar '14

04 Mar '14

The wm8763 driver uses regmap for IO which means that codec->cache_sync is not used. The line was added in commit d3398ff ('ASoC: Convert WM8753 to direct regmap API usage'). Presumably this was meant to be regcache_mark_dirty(), but since we already call regcache_mark_dirty() in the core when suspending the CODEC it is safe to just remove the line. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8753.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da9..a02e76c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1440,7 +1440,6 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - codec->cache_sync = 1; return 0; } -- 1.8.0

3 2

[alsa-devel] change kmalloc into vmalloc for large memory allocations
by Wang, Yalin 04 Mar '14

04 Mar '14

Hi I find there is some drivers use kmalloc to allocate large Memorys during module_init, some can be changed to use vmalloc To save some low mem, I add log in kernel to track , And list them here: https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/us… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/sound/soc/… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/net/netfil… https://git.kernel.org/cgit/linux/kernel/git/will/linux.git/tree/drivers/in… they allocate large memory from 10k~64K , this will use lots of low mem, instead if we use vmalloc for these drivers , it will be good for some devices like smart phone, we often encounter some errors like kmalloc failed because there is not enough low mem , especially when the device has physical memory less than 1GB . could this module change the memory allocation into vmalloc ? I was thinking that if we can introduce a helper function in kmalloc.h like this : Kmalloc(size, flags) { If (size > PAGE_SIZE && flags&CAN_USE_VMALLOC_FLAG) return vmalloc(size); Else return real_kmalloc(size); } Kfree(ptr) { If (is_vmalloc_addr(ptr)) Vfree(ptr); Else Kfree(ptr); } But we need add some flags to ensure always use kmalloc for Some special use (dma etc..) How do you think of it ? Thanks

7 18

[alsa-devel] [PATCH] ASoC: snappercl15: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7..bdc88e5 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int snappercl15_tlv320aic23_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_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", @@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-i2s", - .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, @@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = { .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int snappercl15_probe(struct platform_device *pdev) -- 1.8.0

2 2

[alsa-devel] [PATCH] sound: omap: n810: fix init with DT boot
by Aaro Koskinen 03 Mar '14

03 Mar '14

Since 3.14-rc1 only DT boot has been supported on N810. Make a minimal fix to retain functionality. This file should be properly converted to DT in longer term. Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> --- sound/soc/omap/n810.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e4..d163e18 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -305,7 +305,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); -- 1.8.3.3

2 1

[alsa-devel] Enable headset Mic for Dell DT
by Kailang 03 Mar '14

03 Mar '14

Hi Takashi, I created patch below for Dell DT for support headset MIC. Many Thanks. Kailang. >From 2f448d6f9ad7023befa91b80f86cb267311d3f8a Mon Sep 17 00:00:00 2001 From: Kailang Yang <kailang(a)realtek.com> Date: Mon, 3 Mar 2014 15:27:22 +0800 Subject: [PATCH] ALSA: hda/realtek - Add headset quirk for Dell DT This quirk is needed for the headset microphone to work. Signed-off-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..74456bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3594,21 +3594,38 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc255_set_default_jack_type(struct hda_codec *codec) +{ + /* Set to iphone type */ + alc_write_coef_idx(codec, 0x1b, 0x880b); + alc_write_coef_idx(codec, 0x45, 0xd089); + alc_write_coef_idx(codec, 0x1b, 0x080b); + alc_write_coef_idx(codec, 0x46, 0x0004); + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + msleep(30); +} + static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_PRE_PROBE) { - /* Set to iphone type */ - alc_write_coef_idx(codec, 0x1b, 0x880b); - alc_write_coef_idx(codec, 0x45, 0xd089); - alc_write_coef_idx(codec, 0x1b, 0x080b); - alc_write_coef_idx(codec, 0x46, 0x0004); - alc_write_coef_idx(codec, 0x1b, 0x0c0b); - msleep(30); + alc255_set_default_jack_type(codec); } alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + alc255_set_default_jack_type(codec); + } + else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3875,7 +3892,9 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, + ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -4248,10 +4267,23 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, }, + [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc255_no_hp_mic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4314,6 +4346,8 @@ 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, 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), 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),

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[alsa-devel] [PATCH] ALSA: hda - Added inverted digital-mic handling for Acer TravelMate 8371
by Marius Knaust 03 Mar '14

03 Mar '14

Signed-off-by: Marius Knaust <marius.knaust(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 534796e..9fc7bb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4255,6 +4255,7 @@ static const struct hda_fixup alc269_fixups[] = { }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Make azx_attach_pcm_stream static
by Dylan Reid 03 Mar '14

03 Mar '14

It is only used in hda_controller.c now. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_controller.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6156d0a..97993e1 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -971,8 +971,8 @@ static void azx_pcm_free(struct snd_pcm *pcm) #define MAX_PREALLOC_SIZE (32 * 1024 * 1024) -int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, - struct hda_pcm *cpcm) +static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, + struct hda_pcm *cpcm) { struct azx *chip = bus->private_data; struct snd_pcm *pcm; -- 1.8.1.3.605.g02339dd

2 3

[alsa-devel] [PATCH] ASoC: wm1133-ev1: Convert to table based DAPM setup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/wm1133-ev1.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fce6325..804749a 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -214,12 +214,6 @@ static int wm1133_ev1_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_new_controls(dapm, wm1133_ev1_widgets, - ARRAY_SIZE(wm1133_ev1_widgets)); - - snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, - ARRAY_SIZE(wm1133_ev1_map)); - /* Headphone jack detection */ snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), @@ -257,6 +251,11 @@ static struct snd_soc_card wm1133_ev1 = { .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, + + .dapm_widgets = wm1133_ev1_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets), + .dapm_routes = wm1133_ev1_map, + .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map), }; static struct platform_device *wm1133_ev1_snd_device; -- 1.8.0

2 1

Re: [alsa-devel] 7.1 surround not working on ALC892/ Ubuntu 13.04
by Raymond Yau 03 Mar '14

03 Mar '14

> I am unable to get 7.1 surround sound option on my system. > Alsamixer is not giving me any option to choose 8 channels. > Surround 5.1 is working correctly. > I am not sure how to configure the system to get the 8 channel setup working. > >Distributor ID: Ubuntu >Description: Ubuntu 13.04 >Release: 13.04 >Codename: raring >Attaching all the sound-card information I could gather from the system. >Please let me know in case any additional information is needed to debug the issue. > user1 <at> user1-MI970VF:~/ufouser$ cat /proc/asound/card0/codec#* | grep Codec > Codec: Realtek ALC892 For motherboard with 3 audio jacks at rear panel and 10 channels HDA codec you need to retask headphone at front panel as side line out the easy way is to use hdajackretask (advanced override) or early patching to retask if green line out at rear panel has sequence 1 in pin default. 1) blue line jack as Line Out with sequence 2 2) pink mic jack as Line Out with sequence 3 3) front panel headphone to Line out with sequence 4 the three retasked line out should have same default association as the green line out in theory, it is possible to add the retask of headphone to line out into the existing channel mode (8ch) when the codec has extra DAC and volume control using multi_Ios pin butt need to hack the jack detection control, disable auto mute and independent headphone when the headphone is retasked

2 1

[alsa-devel] [PATCHv2 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 03 Mar '14

03 Mar '14

4 11

[alsa-devel] [PATCH 1/2] ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown
by Xiang Xiao 03 Mar '14

03 Mar '14

The widgets generated by the machine driver need to power off too. Signed-off-by: Xiang Xiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-dapm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3651a37..00ed0e9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3899,7 +3899,7 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) +static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) { struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; @@ -3939,12 +3939,12 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { - struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - soc_dapm_shutdown_codec(&codec->dapm); - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) - snd_soc_dapm_set_bias_level(&codec->dapm, + list_for_each_entry(dapm, &card->dapm_list, list) { + soc_dapm_shutdown_dapm(dapm); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_OFF); } } -- 1.8.1.2

2 3

[alsa-devel] [PATCHv9][ 1/2] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 03 Mar '14

03 Mar '14

Cc: Eric Bénard <eric(a)eukrea.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v9->v10: - Better Cc list - Some non-arm architecture that have CONFIG_OF could select that driver, but they lacked asm/mach-types.h, so the compilation failed. That has been fixed by permitting, when CONFIG_OF is selected, to select the driver only if CONFIG_ARM is set. ChangeLog v7->v8: - fixed the vendor prefixes in the dts bindings. - updated the code and the documentation accordingly. - removed the need of an audio-codec property, since it's available in the ssi bindings. - the dai codec is now retrived trough the dt in order to support machines that have its codec on another i2c bus number. - driver compatible is now similar to the ones of the other drivers. ChangeLog v6->v7: - The cleanups went into another patch. - The support for fsl_ssi doesn't depend on dt anymore. - platform_name = "imx-ssi.0" is still set when in non-dt mode. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 108 +++++++++++++++++--- 3 files changed, 121 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..0d7985c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,21 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,asoc-tlv320" +- eukrea,model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f14..b8f8703 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,12 +168,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 5983740..eb093d5 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -15,8 +15,11 @@ * */ +#include <linux/errno.h> #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,6 +29,7 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 @@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) { + /* fsl_ssi lacks the set_fmt ops. */ + if (ret && ret != -ENOTSUPP) { dev_err(cpu_dai->dev, "Failed to set the cpu dai format.\n"); return ret; @@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, "Failed to set the codec sysclk.\n"); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); - if (ret) { + /* fsl_ssi lacks the set_sysclk ops */ + if (ret && ret != -EINVAL) { dev_err(cpu_dai->dev, "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; @@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, @@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; - if (machine_is_eukrea_cpuimx27()) { + eukrea_tlv320.dev = &pdev->dev; + if (np) { + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); + if (ret) { + dev_err(&pdev->dev, + "eukrea,model node missing or invalid.\n"); + goto err; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + if (!ssi_np) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + if (codec_np) + eukrea_tlv320_dai.codec_of_node = codec_np; + else + dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-int-port node missing or invalid.\n"); + return ret; + } + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-ext-port node missing or invalid.\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_of_node = ssi_np; + eukrea_tlv320_dai.platform_of_node = ssi_np; + } else { + eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0"; + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a"; + eukrea_tlv320.name = "cpuimx-audio"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,asoc-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing or invalid audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } - eukrea_tlv320.dev = &pdev->dev; ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,asoc-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove, -- 1.7.9.5

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[alsa-devel] [PATCH 1/8] ASoC: pxa: Pass correct DAPM context to {corgi, poodle, spitz}_ext_control
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

When calling {corgi,poodle,spitz}_ext_control() from the startup callback we pass the CODEC's DAPM context instead of the card's DAPM context. This is not a problem per se since all the DAPM functions in ext_control() fallback to widgets from other DAPM contexts, but passing the card's context is more consistent. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 3 +-- sound/soc/pxa/poodle.c | 3 +-- sound/soc/pxa/spitz.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b4db3b9..5a88136 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -101,10 +101,9 @@ static void corgi_ext_control(struct snd_soc_dapm_context *dapm) static int corgi_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - corgi_ext_control(&codec->dapm); + corgi_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 27c6c03..c6bdc6c 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -74,10 +74,9 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) static int poodle_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - poodle_ext_control(&codec->dapm); + poodle_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index a3a13c9..1373b01 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -111,10 +111,9 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) static int spitz_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - spitz_ext_control(&codec->dapm); + spitz_ext_control(&rtd->card->dapm); return 0; } -- 1.8.0

2 16

[alsa-devel] [PATCH 1/2] ASoC: neo1973_wm8753: Post gta01 support removal cleanup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

With GTA01 support gone from the driver there is no need to keep a separate initilization routine for the GTA02 specific bits. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 116 ++++++++++++------------------------- 1 file changed, 37 insertions(+), 79 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 98a04c1..0125d9d 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -/* Shared routes and controls */ - -static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), -}; - -static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), -}; - -/* GTA02 specific routes and controls */ - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { +static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), +}; + +static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, + /* Connections to the amp */ {"Stereo Out", NULL, "LOUT1"}, {"Stereo Out", NULL, "ROUT1"}, @@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { {"Handset Spk", NULL, "ROUT2"}, }; -static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { +static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), SOC_DAPM_PIN_SWITCH("Handset Spk"), SOC_DAPM_PIN_SWITCH("Stereo Out"), @@ -276,39 +269,6 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { lm4853_set_spk), }; -static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), -}; - -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets, - ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes, - ARRAY_SIZE(neo1973_gta02_routes)); - if (ret) - return ret; - - ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls, - ARRAY_SIZE(neo1973_gta02_wm8753_controls)); - if (ret) - return ret; - - snd_soc_dapm_disable_pin(dapm, "Stereo Out"); - snd_soc_dapm_disable_pin(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - - return 0; -} - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -344,18 +304,16 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "GSM Line In"); snd_soc_dapm_disable_pin(dapm, "Headset Mic"); snd_soc_dapm_disable_pin(dapm, "Handset Mic"); + snd_soc_dapm_disable_pin(dapm, "Stereo Out"); + snd_soc_dapm_disable_pin(dapm, "Handset Spk"); /* allow audio paths from the GSM modem to run during suspend */ snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); - - if (machine_is_neo1973_gta02()) { - ret = neo1973_gta02_wm8753_init(codec); - if (ret) - return ret; - } + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); return 0; } -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 02 Mar '14

02 Mar '14

dpcm_path_get() allocates dynamic memory to hold path list. Corresponding dpcm_get_path() must be called to free the memory. dpcm_get_path() is not called under several error conditions. This leads to memory leak. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ 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); @@ -2018,6 +2019,7 @@ 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); @@ -2082,6 +2084,7 @@ 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"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

1 0

[alsa-devel] [RFC 00/19] Enable platform HDA drivers
by Dylan Reid 01 Mar '14

01 Mar '14

This series rearranges some code in the HDA driver to break dependencies on PCI and allow for most of the HDA code to be reused for non-PCI based HDA controllers. I tried to break it up to make it less scary and to keep changes away from copies of code. It is mostly copies with a few modifications to make more of the code usable by both HDA drivers. The main changes were to avoid using pci to get a device pointer, and to make the bus reads/writes into callbacks so that the Tegra driver could ensure 32-bit access to the HDA registers. The final patch is a work in progress included here to provide context for the preceding patches. The Tegra HDA driver still needs some cleaning up, but nothing that will affect the other changes. The last infrastructure issue is that the Tegra driver uses DT, causing it to be probed before the codec patches. I haven't determined how to fix that yet. I didn't include the patch to move the hda directory from under pci because of its size and how easy it will be to regenerate later. Thanks for looking. Let me know if you think any of these should be squashed or re-ordered and if the changes and breakup of code makes sense. Change since v1: - Rename hda_shared to hda_controller - Add more 'ops' to break out hda_intel specific functions, such as postponed interrupt handling and page allocation - Re order a few patches to make rebases easier - Make bdl_pos_adj optional - I left the position fix code in hda_controller, I don't have a good way to test a refactoring of that, there are a few combinations of settings. It will probably be useful to other controllers as well, but moving it wouldn't be hard if we really want to. Thanks again, Dylan -- Dylan Reid (19): ALSA: hda - Move some definitions to new hda_priv.h ALSA: hda - Allow different ops to read/write registers ALSA: hda - Keep pointer to bdl_pos_fix in chip struct ALSA: hda - Store device pointer in chip structure ALSA: hda - Move pcm ops and support funcs to shared file ALSA: hda - Pull pages allocation to shared file ALSA: hda - Move the dsp loader to hda_shared ALSA: hda - Add function pointer for disabling MSI ALSA: hda - Relocate RIRB/CORB interface to hda_shared ALSA: hda - move alloc_cmd_io to hda_shared. ALSA: hda - Move low level functions to hda_shared ALSA: hda - remove unused clear of STATESTS ALSA: hda - Move azx_interrupt to hda_shared ALSA: hda - Add jackpoll_ms to struct azx ALSA: hda - Pass max_slots and power_save to codec_create ALSA: hda - Move codec create to hda_shared ALSA: core - Define snd_pci_quirk without CONFIG_PCI ALSA: hda - remove PCI dependency in Kconfig WIP: ALSA: hda - Add driver for Tegra SoC HDA include/sound/core.h | 15 +- sound/pci/Kconfig | 4 +- sound/pci/hda/Kconfig | 55 +- sound/pci/hda/Makefile | 8 +- sound/pci/hda/hda_intel.c | 2832 +++----------------------------------------- sound/pci/hda/hda_priv.h | 446 +++++++ sound/pci/hda/hda_shared.c | 2175 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_shared.h | 52 + sound/pci/hda/hda_tegra.c | 858 ++++++++++++++ 9 files changed, 3778 insertions(+), 2667 deletions(-) create mode 100644 sound/pci/hda/hda_priv.h create mode 100644 sound/pci/hda/hda_shared.c create mode 100644 sound/pci/hda/hda_shared.h create mode 100644 sound/pci/hda/hda_tegra.c -- 1.8.1.3.605.g02339dd

2 25

[alsa-devel] [PATCH 0/3] To simplify the code for ASoC probe.
by Xiubo Li 01 Mar '14

01 Mar '14

Xiubo Li (3): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: SGTL5000: Simplify ASoC probe code. sound/soc/codecs/sgtl5000.c | 8 -------- sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-io.c | 5 ----- 3 files changed, 4 insertions(+), 17 deletions(-) -- 1.8.4

4 18

[alsa-devel] [PATCH 14/28] Remove MACH_SMDKC210
by Richard Weinberger 01 Mar '14

01 Mar '14

The symbol is an orphan, get rid of it. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41c..7ea576c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.4.2

3 9

[alsa-devel] [PATCH v2 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 01 Mar '14

01 Mar '14

Changes since v1: * Rename the xmax parameter for the SOC_ENUM_SINGLE_VIRT() macro to xitems. Original cover letter below. We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm899 Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

2 12

[alsa-devel] Audio passthrough on Pandaboard using ALSA API
by Vadim Izmalkov 28 Feb '14

28 Feb '14

Hello I created simple application that makes audio pass-through (from analog input to analog output), After 2-10 seconds get error like: AlsaPassthrough: pcm.c:693: snd_pcm_close: Assertion `pcm' failed. or write to audio interface failed (Broken pipe) Seems I have problem with buffer size/time or somethink like that. Parameters used: Buffer - 100 ms / 4800 frames / 19200 bytes Period - 10 ms / 480 frames / 1920 bytes Buffer length used 19200 bytes. char buffer[19200]; Write/read function used as follow snd_pcm_readi(pcm, buffer, frames); snd_pcm_writei(pcm, buffer, frames); Have somebody any suggestion? Segments of the code you can find bellow. Alsamixer settings: amixer cset name='Analog Left Capture Route' 'Headset Mic' amixer cset name='Analog Right Capture Route' 'Headset Mic' amixer cset name='MUX_UL00' 'AMic0' amixer cset name='MUX_UL01' 'AMic1' amixer cset name='Headset Left Playback' 'HS DAC' amixer cset name='Headset Right Playback' 'HS DAC' amixer cset name='Headset Playback Volume' 100 amixer cset name='DL1 Media' 120 Structure for the devices: typedef struct { char *deviceName; snd_pcm_t *pcm; snd_pcm_hw_params_t *hwParams; snd_pcm_stream_t stream; snd_pcm_access_t access; snd_pcm_format_t format; snd_pcm_uframes_t frames; int mode; unsigned int sampleRate; unsigned int bufferTime; unsigned int periodTime; unsigned int channels; } pcmDevice_t; Parameters for capture: pcmDevice_t* initRecordDevice() { pcmDevice_t *device; device = malloc(sizeof(pcmDevice_t)); device->deviceName = strdup("plughw:0,0"); device->stream = SND_PCM_STREAM_CAPTURE; device->access = SND_PCM_ACCESS_RW_INTERLEAVED; device->format = SND_PCM_FORMAT_S16_LE; device->mode = 0; //SND_PCM_ASYNC; //SND_PCM_NONBLOCK; device->sampleRate = 48000; device->bufferTime = 100 * 1000; // 100 ms device->periodTime = 10 * 1000; // 20ms device->channels = 2; return device; } Parameters for playback: pcmDevice_t* initPlaybackDevice() { pcmDevice_t *device; device = malloc(sizeof(pcmDevice_t)); device->deviceName = strdup("plughw:0,0"); device->stream = SND_PCM_STREAM_PLAYBACK; device->access = SND_PCM_ACCESS_RW_INTERLEAVED; device->format = SND_PCM_FORMAT_S16_LE; device->mode = 0; //SND_PCM_ASYNC; //SND_PCM_NONBLOCK; device->sampleRate = 48000; device->bufferTime = 100 * 1000; // 100 ms device->periodTime = 10 * 1000; // 20ms device->channels = 2; return device; } Setup devices: int setupDevice(pcmDevice_t *device) { int err; /* Open PCM. The last parameter of this function is the mode. */ /* If this is set to 0, the standard mode is used. Possible */ /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */ /* If SND_PCM_NONBLOCK is used, read / write access to the */ /* PCM device will return immediately. If SND_PCM_ASYNC is */ /* specified, SIGIO will be emitted whenever a period has */ /* been completely processed by the soundcard. */ if ((err = snd_pcm_open(&device->pcm, device->deviceName, device->stream, device->mode)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", device->deviceName, snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_malloc (&device->hwParams)) < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_any (device->pcm, device->hwParams)) < 0) { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_access (device->pcm, device->hwParams, device->access)) < 0) { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_format (device->pcm, device->hwParams, device->format)) < 0) { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_buffer_time_near (device->pcm, device->hwParams, &device->bufferTime, NULL)) < 0) { fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)); exit (1); } else { /* Set buffer size (in frames). The resulting latency is given by */ /* latency = periodsize * periods / (rate * bytes_per_frame) */ snd_pcm_hw_params_get_buffer_size(device->hwParams, &device->frames); } if ((err = snd_pcm_hw_params_set_period_time_near (device->pcm, device->hwParams, &device->periodTime, NULL)) < 0) { fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_rate_near (device->pcm, device->hwParams, &device->sampleRate, NULL)) < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)); exit (1); } if ((err = snd_pcm_hw_params_set_channels (device->pcm, device->hwParams, device->channels)) < 0) { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)); exit (1); } /* Apply HW parameter settings to */ /* PCM device and prepare device */ if ((err = snd_pcm_hw_params (device->pcm, device->hwParams)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); exit (1); } snd_pcm_hw_params_free (device->hwParams); if ((err = snd_pcm_prepare (device->pcm)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } return 0; } Main: main (int argc, char *argv[]) { int i; int err; char buf[4800*4]; snd_pcm_t *capture_handle; snd_pcm_hw_params_t *hw_params; snd_pcm_uframes_t frames; pcmDevice_t *pcmRecord; pcmDevice_t *pcmPlayback; pcmRecord = initRecordDevice(); pcmPlayback = initPlaybackDevice(); printf("\n----------- RECORD DEVICE -----------\n"); setupDevice(pcmRecord); showConfig(pcmRecord); printf("\n---------- PLAYBACK DEVICE ----------\n"); setupDevice(pcmPlayback); showConfig(pcmPlayback); for (i = 0; i < 100; ++i) { /* Read num_frames frames from the PCM device */ /* pointed to by pcm_handle to buffer capdata. */ /* Returns the number of frames actually read. */ //if((err = snd_pcm_readi (pcmRecord->pcm, buf, pcmRecord->frames)) != pcmRecord->frames) { if((err = snd_pcm_readi (pcmRecord->pcm, buf, 480)) != 480) { fprintf (stderr, "read from audio interface failed (%s)\n", snd_strerror (err)); exit (1); } /* Write num_frames frames from buffer data to */ /* the PCM device pointed to by pcm_handle. */ /* Returns the number of frames actually written. */ if((err = snd_pcm_writei (pcmPlayback->pcm, buf, pcmRecord->frames)) != pcmRecord->frames) { fprintf (stderr, "write to audio interface failed (%s)\n", snd_strerror (err)); exit (1); } } snd_pcm_close (capture_handle); exit (0); }? Best

2 1

[alsa-devel] [alsa-plugins] [PATCH 1/1] Allow number of channels
by Corfu, ValentinX 28 Feb '14

28 Feb '14

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1 0

Re: [alsa-devel] Griffin FireWave stream format information
by Stefan Richter 28 Feb '14

28 Feb '14

On Feb 27 Takashi Sakamoto wrote: > Stefan or Jay, > > Could I request you to get outputs against command in this message with > FireWave? Jay already responded, but [...] > Based on this guess and according to its manual, FireWave should have 7 > entries: > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff01ff > 0c ff bf c1 00 00 00 00 ff 00 01 90 40 02 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff01ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 01 90 40 02 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff02ff > 0c ff bf c1 00 00 00 00 ff 00 02 90 40 03 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff02ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 02 90 40 03 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff03ff > 0c ff bf c1 00 00 00 00 ff 00 03 90 40 04 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff03ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 03 90 40 05 02 01 ............@... response: 010: 06 06 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff04ff > 0c ff bf c1 00 00 00 00 ff 00 04 90 40 05 02 01 06 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff04ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 04 90 40 02 02 01 ............@... response: 010: 02 00 .. > (These four is for 6ch sound for usual 5.1ch surround) > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff05ff > 0c ff bf c1 00 00 00 00 ff 00 05 90 40 02 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff05ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 05 90 40 03 02 01 ............@... response: 010: 02 00 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff06ff > 0c ff bf c1 00 00 00 00 ff 00 06 90 40 03 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff06ff response: 000: 0c ff bf c1 00 00 00 00 ff 00 06 90 40 04 02 01 ............@... response: 010: 02 00 .. > $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff07ff > 0c ff bf c1 00 00 00 00 ff 00 07 90 40 04 02 01 02 06 $ firewire-request /dev/fw8 fcp 0x01ffbfc100000000ffff07ff response: 000: 0a ff bf c1 00 00 00 00 ff ff 07 ff ............ > (These three is for 2ch stereo sound for Dolby Pro Logic II) > > Would you please confirm these output? So this seems to be the same as from Jay's FireWave specimen. Here are results of the other requests from your subsequent mail: $ firewire-request /dev/fw8 fcp 0x01ffbfc000000000ffffffff response: 000: 0c ff bf c0 00 00 00 00 ff 00 90 40 04 02 01 06 ...........@.... response: 010: 06 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400202010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 02 02 01 02 ...........@.... response: 010: 00 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400302010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 03 02 01 02 ...........@.... response: 010: 00 . $ firewire-request /dev/fw8 fcp 0x00ffbfc100000000ff0090400402010200 response: 000: 08 ff bf c1 00 00 00 00 ff 00 90 40 04 02 01 02 ...........@.... response: 010: 00 . -- Stefan Richter -=====-====- --=- ===-- http://arcgraph.de/sr/

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[alsa-devel] [PATCH v2 0/3] alsa-lib: Improve 2.1 surround support
by David Henningsson 28 Feb '14

28 Feb '14

Thanks Takashi for your encouraging review, Diff since last version is only the things you remarked about in the first patch series. > Besides that, I find it also OK to just return the channel (>= 0) or a > negative error. But it's no big matter in which form is. This one I didn't change, just because then I would just have had to change it back in the second patch anyway. David Henningsson (3): route: Allow chmap syntax for slave channels in ttable route: Select slave chmap based on ttable information conf: Allow 2.1 surround to use different number of channels src/conf/pcm/surround21.conf | 7 +- src/pcm/pcm_route.c | 323 +++++++++++++++++++++++++++++++++++++----- 2 files changed, 290 insertions(+), 40 deletions(-) -- 1.7.9.5

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[alsa-devel] hda: Bug - (timing?) problem causes DP monitor to not get activated
by Daniel Martin 28 Feb '14

28 Feb '14

Hi, I'm having a bug here, where the loaded hda module causes some (timing?) problem resulting in a monitor connected to the DP not to get active. I can reproduce this on a Lenovo X201 and T410 (Ironlake) sitting on a docking station, which has a monitor (Dell U2410) connected to a DP. Steps to reproduce: 1. boot with the laptop at the docking station and the monitor plugged in 2. start the X-Server ... output on LVDS and DP is okay 3. disable the DP: xrandr --output LVDS1 --preferred --output DP1 --off ... output on LVDS - okay 4. switch to DP: xrandr --output LVDS1 --off --output DP1 --preferred ... DP _usually_ doesn't gets active, it turn off into suspend state Why I think that it is related to the hda module? If I boot a kernel without the hda module step 4 always works. When having the hda module loaded I need some luck and some iterations of step 3 and 4 to activate the monitor at the DP. I verified that the problem exists in kernel v3.13.1 and v3.14-rc4. The logs I have attached have been created on the T410 with kernel v3.14-rc4. The first alsa-info....txt has been created right after booting, the second one after step 4 failed. The kernel has been build with all alsa debug options I could find, the config is attached as well, and the kernel has been booted with drm.debug=0xe - this might be to much, but it shows pretty clear when the outputs have been switched. I hope that anyone can read something usefull - something that went wrong - out of the log files. My unexercised eye can't spot anything obvious. If more logs are necessary, I'll send them. Cheers, Daniel Martin

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[alsa-devel] [PATCH] ALSA: hda - Fix registration of beep input device
by Takashi Iwai 28 Feb '14

28 Feb '14

The beep input device is registered via input_register_device(), but this is called in snd_hda_attach_beep_device() where the sound devices aren't registered yet. This leads to the binding to non-existing object, thus results in failure. And, even if the binding worked (against the PCI object), it's still racy; the input device appears before the sound objects. For fixing this, register the input device properly at dev_register ops of the codec object it's bound with. Also, call snd_hda_detach_beep_device() at dev_disconnection so that it's detached at the right timing. As a bonus, since it's called in the codec's ops, we can get rid of the further call from the other codec drivers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 27 ++++++++++++++++++++------- sound/pci/hda/hda_beep.h | 5 +++++ sound/pci/hda/hda_codec.c | 7 ++++++- sound/pci/hda/hda_generic.c | 1 - sound/pci/hda/patch_conexant.c | 4 +--- 5 files changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 88bb08486f77..b9d3d0cd75c9 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -148,7 +148,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) { struct input_dev *input_dev; struct hda_codec *codec = beep->codec; - int err; input_dev = input_allocate_device(); if (!input_dev) @@ -169,12 +168,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->dev.parent = &codec->dev; input_set_drvdata(input_dev, beep); - err = input_register_device(input_dev); - if (err < 0) { - input_free_device(input_dev); - codec_err(codec, "hda_beep: unable to register input device\n"); - return err; - } beep->dev = input_dev; return 0; } @@ -244,6 +237,26 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); +int snd_hda_register_beep_device(struct hda_codec *codec) +{ + struct hda_beep *beep = codec->beep; + int err; + + if (!beep || !beep->dev) + return 0; + + err = input_register_device(beep->dev); + if (err < 0) { + codec_err(codec, "hda_beep: unable to register input device\n"); + input_free_device(beep->dev); + codec->beep = NULL; + kfree(beep); + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_register_beep_device); + static bool ctl_has_mute(struct snd_kcontrol *kcontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index cb88464676b6..76eb8589fb9e 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -45,6 +45,7 @@ struct hda_beep { int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); +int snd_hda_register_beep_device(struct hda_codec *codec); #else static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { @@ -53,5 +54,9 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } +static inline int snd_hda_register_beep_device(struct hda_codec *codec) +{ + return 0; +} #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6db2dbcbf4d3..555e8c771cfd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1379,14 +1379,19 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, static int snd_hda_codec_dev_register(struct snd_device *device) { struct hda_codec *codec = device->device_data; + int err = device_add(&codec->dev); - return device_add(&codec->dev); + if (err < 0) + return err; + snd_hda_register_beep_device(codec); + return 0; } static int snd_hda_codec_dev_disconnect(struct snd_device *device) { struct hda_codec *codec = device->device_data; + snd_hda_detach_beep_device(codec); device_del(&codec->dev); return 0; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 9e0609a4b2ba..16133881e967 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5350,7 +5350,6 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_init); void snd_hda_gen_free(struct hda_codec *codec) { snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE); - snd_hda_detach_beep_device(codec); snd_hda_gen_spec_free(codec->spec); kfree(codec->spec); codec->spec = NULL; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 09d6d0db2b06..1dc7e974f3b1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -445,9 +445,7 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_hda_detach_beep_device(codec); - kfree(spec); + kfree(codec->spec); } static const struct snd_kcontrol_new cxt_capture_mixers[] = { -- 1.9.0

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Re: [alsa-devel] PCM rules for FireWave in speaker.c
by Clemens Ladisch 28 Feb '14

28 Feb '14

Takashi Sakamoto wrote: > Current speaker.c have a rule for PCM channels/rates below: > - 6ch PCM at 32.0/44.1kHz > - 2ch PCM at 48.0/96.0kHz The current rules in speakers.c disallow rates less than 48 kHz if there are less than six channels, and disallow two channels if the rate is less than 48 kHz. This means that the allowed parameter combinations are 2ch at 48/96 kHz, and 6ch at 32/44.1/48/96 kHz. > Recently, with Jay Fenlason's help, I got some information about its > stream format. I'll show them in the end of this message. > > As a result, I realized FireWave has stream formats below: > - 6ch PCM at 32.0/44.1/48.0/96.0kHz > - 2ch IEC 60958-3 at 32.0/44.1/48.0kHz > > Although there are no stream formats for 2ch PCM at 48.0/96.0kHz, the > driver has it. Do you know the reason? Are there any model quirks? I don't have any of the AV/C audio specifications; these rules were determined experimentally. In other words, when the sample rate is set to 48 or 96 kHz, sending a stream with two PCM channels appears to work. If you wanted to support these undocumented formats in your driver, you would have to add some kind of quirk. Regards, Clemens

2 1

[alsa-devel] [PATCH 0/2] Simplify the CODEC ASoC probe code.
by Xiubo Li 28 Feb '14

28 Feb '14

3 4

[alsa-devel] [RFC 00/19] Enable platform HDA drivers
by Dylan Reid 28 Feb '14

28 Feb '14

This series rearranges some code in the HDA driver to break dependencies on PCI and allow for most of the HDA code to be reused for non-PCI based HDA controllers. I tried to break it up to make it less scary and to keep changes away from copies of code. It is mostly copies with a few modifications to make more of the code usable by both HDA drivers. The main changes were to avoid using pci to get a device pointer, and to make the bus reads/writes into callbacks so that the Tegra driver could ensure 32-bit access to the HDA registers. The final patch is a work in progress included here to provide context for the preceding patches. The Tegra HDA driver still needs some cleaning up, but nothing that will affect the other changes. The last infrastructure issue is that the Tegra driver uses DT, causing it to be probed before the codec patches. I haven't determined how to fix that yet. I didn't include the patch to move the hda directory from under pci because of its size and how easy it will be to regenerate later. Thanks for looking. Let me know if you think any of these should be squashed or re-ordered and if the changes and breakup of code makes sense. Dylan -- Dylan Reid (19): ALSA: hda - Move some definitions to new hda_priv.h ALSA: hda - Allow different ops to read/write registers ALSA: hda - Keep pointer to bdl_pos_fix in chip struct ALSA: hda - Store device pointer in chip structure ALSA: hda - Move pcm ops and support funcs to shared file ALSA: hda - Pull pages allocation to shared file ALSA: hda - Move the dsp loader to hda_shared ALSA: hda - Add function pointer for disabling MSI ALSA: hda - Relocate RIRB/CORB interface to hda_shared ALSA: hda - move alloc_cmd_io to hda_shared. ALSA: hda - Move low level functions to hda_shared ALSA: hda - remove unused clear of STATESTS ALSA: hda - Move azx_interrupt to hda_shared ALSA: hda - Add jackpoll_ms to struct azx ALSA: hda - Pass max_slots and power_save to codec_create ALSA: hda - Move codec create to hda_shared ALSA: core - Define snd_pci_quirk without CONFIG_PCI ALSA: hda - remove PCI dependency in Kconfig WIP: ALSA: hda - Add driver for Tegra SoC HDA include/sound/core.h | 15 +- sound/pci/Kconfig | 4 +- sound/pci/hda/Kconfig | 55 +- sound/pci/hda/Makefile | 8 +- sound/pci/hda/hda_intel.c | 2832 +++----------------------------------------- sound/pci/hda/hda_priv.h | 446 +++++++ sound/pci/hda/hda_shared.c | 2175 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_shared.h | 52 + sound/pci/hda/hda_tegra.c | 858 ++++++++++++++ 9 files changed, 3778 insertions(+), 2667 deletions(-) create mode 100644 sound/pci/hda/hda_priv.h create mode 100644 sound/pci/hda/hda_shared.c create mode 100644 sound/pci/hda/hda_shared.h create mode 100644 sound/pci/hda/hda_tegra.c -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 28 Feb '14

28 Feb '14

We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm8995 - Lars Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

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[alsa-devel] [RFC PATCH] ALSA: hda - Fix CORB reset to follow specification
by David Henningsson 28 Feb '14

28 Feb '14

According to the HDA spec, we must write 1 to bit 15 on a CORBRP reset, read back 1, then write 0, then read back 0. This must be done while the DMA is not running. We accidentaly ended up writing back the 0 by using a writel instead of a writew to CORBWP. This caused occasional controller failure on Bay Trail hardware. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) Hi Takashi, A while ago I researched a spurious response on some Bay Trail hardware, so I proofread some of our CORB/RIRB code and found the below. When applied, the spurious responses disappeared but other problems might have appeared instead. I don't have the hardware here so I'm not 100% sure. So I'm hesitant to actually apply it, especially as the existing non-compliant way seems to work for all other chips. Still wanted to show the issue to you. Also the timeout (1 ms) is completely arbitrarily as there are no time limits written in the HDA specification. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8d9bf3..59156f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -775,6 +775,8 @@ static int azx_alloc_cmd_io(struct azx *chip) static void azx_init_cmd_io(struct azx *chip) { + int timeout; + spin_lock_irq(&chip->reg_lock); /* CORB set up */ chip->corb.addr = chip->rb.addr; @@ -786,8 +788,28 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, CORBSIZE, 0x02); /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); + /* reset the corb hw read pointer */ azx_writew(chip, CORBRP, ICH6_CORBRP_RST); + for (timeout = 1000; timeout > 0; timeout--) { + if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout - CORBRP = %d\n", + pci_name(chip->pci), (int) azx_readw(chip, CORBRP)); + + azx_writew(chip, CORBRP, 0); + for (timeout = 1000; timeout > 0; timeout--) { + if (azx_readw(chip, CORBRP) == 0) + break; + udelay(1); + } + if (timeout <= 0) + snd_printk(KERN_ERR SFX "%s: CORB reset timeout, CORBRP = %d\n", + pci_name(chip->pci), azx_readw(chip, CORBRP)); + /* enable corb dma */ azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); @@ -862,7 +884,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) chip->rirb.cmds[addr]++; chip->corb.buf[wp] = cpu_to_le32(val); - azx_writel(chip, CORBWP, wp); + azx_writew(chip, CORBWP, wp); spin_unlock_irq(&chip->reg_lock); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 28 Feb '14

28 Feb '14

At Thu, 27 Feb 2014 15:32:53 +0800, Raymond Yau wrote: > > 2014-02-26 23:53 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > > > Hi, > > > > this is another long series of patches for trivial cleanups. > > This time, they replace the printk and snd_print*() calls in ALSA PCI > > and USB drivers with dev_err() and co. This will give nicer and more > > consistent outputs. > > > > The only drawback by these replacements is that alsa-info.sh won't be > > able to catch the dmesg lines because it doesn't contain any longer > > the word "ALSA", but only show the module name like "snd_intel8x0". > > > > I'm going to fix alsa-info.sh on git soon later, but if you guys think > > "oh no it's disaster!", let me know. We need to reconsider a better > > way to put the known markers (like explicitly putting "ALSA" at the > > beginning.) > > > > > seem missing au88x0 Yes, because some driver codes are horrible to work on. Patches are welcome. > can the driver build as modules in kernel 3.11 (ubuntu 13.10) ? The dev_err() and co exist since ages ago, so it shouldn't be a problem. These patches, however, need the previous change of snd_card_new() to make the device pointer available in time. No idea about Ubuntu. Takashi

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[alsa-devel] [PATCH] ASoC: simple-card: cpu_dai_name creates confusion when DT case
by Kuninori Morimoto 28 Feb '14

28 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, soc_bind_dai_link() checks cpu_dai->dev->of_node and dai_link->cpu_of_node in DT case. But after that it will check cpu_dai->name and dai_link->cpu_dai_name too. On the other hand, snd_soc_dai :: name is created by fmt_single_name() or fmt_multiple_name(). There is no confusion if dai name is created by fmt_multiple_name(), since cpu_dai->name is same as dai_link->cpu_dai_name. but, if dai name is created by fmt_single_name(), CPU DAI never match. Thus, simple-card not set dai_link->cpu_dai_name if DT case to skip naming match on soc_bind_dai_link() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..4a0e06e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -242,7 +242,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; + if (!of_cpu) + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] ASoC: ARM Cirrus Logic CLPS711X DAI driver
by Alexander Shiyan 28 Feb '14

28 Feb '14

This series adds sound support for the Cirrus Logic CLPS711X CPUs. Driver is designed (and tested) for using with generic sound card. Alexander Shiyan (2): ASoC: Add driver for CLPS711X DAI interface ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI .../bindings/sound/cirrus,clps711x-dai.txt | 20 + sound/soc/cirrus/Kconfig | 8 + sound/soc/cirrus/Makefile | 5 + sound/soc/cirrus/clps711x-dai.c | 522 +++++++++++++++++++++ sound/soc/cirrus/clps711x-dai.h | 61 +++ sound/soc/cirrus/clps711x-fiq.S | 56 +++ 6 files changed, 672 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt create mode 100644 sound/soc/cirrus/clps711x-dai.c create mode 100644 sound/soc/cirrus/clps711x-dai.h create mode 100644 sound/soc/cirrus/clps711x-fiq.S -- 1.8.3.2

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[alsa-devel] [PATCH v2] ASOC: dapm: add params_select callback
by nikesh＠opensource.wolfsonmicro.com 28 Feb '14

28 Feb '14

From: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional callback to select one of the param from a list of params. Signed-off-by: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> --- Change since v1: Used an index to select from a list of params, instead of overwriting the param. Thanks, Nikesh include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 4 files changed, 25 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 68d92e3..12b0d9e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -416,7 +416,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..20110aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -876,6 +876,9 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + /* optional params selection for dai links */ + int (*params_select)(const struct snd_soc_pcm_stream *params, + unsigned int *param_index, int event); unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..4f03e88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1469,7 +1469,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1481,7 +1481,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..2e2f633 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3379,10 +3379,12 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; const struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_dai_link *dai_link = w->priv; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; - int ret; + int ret = 0; + unsigned int param_index; if (WARN_ON(!config) || WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) @@ -3402,6 +3404,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + if (dai_link && dai_link->params_select) { + ret = dai_link->params_select(config, &param_index, event); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: params_select for dai link widget failed %d\n", + ret); + goto out; + } + if (param_index > 0) + config += param_index; + } + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3485,7 +3499,8 @@ out: int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link) { struct snd_soc_dapm_route routes[2]; struct snd_soc_dapm_widget template; @@ -3517,6 +3532,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; + w->priv = (void *)dai_link; memset(&routes, 0, sizeof(routes)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: Separation of the main audio options and boards
by Alexander Shiyan 28 Feb '14

28 Feb '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 07f8f14..ecc1092 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 SND_SOC_GENERIC_DMAENGINE_PCM @@ -14,13 +16,27 @@ 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 @@ -29,6 +45,8 @@ 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 @@ -106,13 +124,6 @@ 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 @@ -122,12 +133,7 @@ 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" @@ -219,3 +225,5 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +endmenu -- 1.8.3.2

2 7

[alsa-devel] [PATCH 1/3] ALSA: hda - Remove dependency on bus->pci in hda_beep.c
by Takashi Iwai 27 Feb '14

27 Feb '14

The default parent device can be obtained directly via card object, so we don't need to rely on pci->dev.parent. Since there is no access to pci_dev, we can reduce the inclusion of linux/pci.h, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0589b39cda6e..d8c437a94559 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -20,7 +20,6 @@ */ #include <linux/input.h> -#include <linux/pci.h> #include <linux/slab.h> #include <linux/workqueue.h> #include <linux/export.h> @@ -167,7 +166,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->bus->pci->dev; + input_dev->dev.parent = codec->bus->card->dev; input_set_drvdata(input_dev, beep); err = input_register_device(input_dev); -- 1.8.5.2

3 13

[alsa-devel] NUC D34010WYK HDMI Audio does not work...
by Robert Krakora 27 Feb '14

27 Feb '14

Hi All, We have a NUC D34010WYK on which I have installed several distros/kernels in a, so far, vain attempt to get HDMI audio to function via ALSA. I upload and alsainfo snap shot here: http://www.alsa-project.org/db/?f=d5f75cecee1195591d85be99c62db09bcd2bbdaf The latest distro I tried was an Ubuntu 14.04 daily from Feburary 18th. I also found this link where someone had tried unsuccessfully to get HDMI audio from this same NUC model unsuccessfully with Ubuntu 13.10: http://ubuntuforums.org/showthread.php?t=2202727 Any and all help will be greatly appreciated! :-) Best Regards, Rob -- Rob Krakora, Senior Software Engineer MessageNet Systems 101 E Carmel Dr, Suite 105 Carmel, IN 46032 MessageNetSystems.com<http://www.messagenetcommunicationsystems.com/?utm_source=email+signature&u…> Rob.Krakora(a)MessageNetSystems.com <rob.krakora(a)messagenetsystems.com> P: 317.566.1677, 212 F: 317.663.0808 For the latest news, information, and blogs, please be sure to visit, follow, and like us... <http://www.messagenetcommunicationsystems.com/get-the-message-out-blog/?utm…> <http://www.youtube.com/user/MessageNetConnection/feed> <http://www.linkedin.com/company/messagenet-systems> <http://twitter.com/MessageNet> <http://www.facebook.com/MessageNetsystems>

2 4

[alsa-devel] [PATCH] ALSA: hda/sigmatel - Allow auto-switching for dock line-in of HP laptops
by Takashi Iwai 27 Feb '14

27 Feb '14

Many HP laptops with STAC codecs have a docking station port and BIOS sets the pins for the input on the dock as a line in. Because the generic parser doesn't handle a line in pin as auto-switchable, this resulted in the manual capture source selection on these laptops. However, from the usability POV, the automatic switching is easier. This patch adds the line_in_auto_switch hint in the fixup function for these laptops. Even if no dock port is present, this should be harmless as the generic parser allows the auto-switching only in a limited situation (all three pins are located in different positions). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3784808758e..75515b494034 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2089,6 +2089,9 @@ static void stac92hd83xxx_fixup_hp(struct hda_codec *codec, codec_dbg(codec, "mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); + + /* allow auto-switching of dock line-in */ + spec->gen.line_in_auto_switch = true; } static void stac92hd83xxx_fixup_hp_zephyr(struct hda_codec *codec, -- 1.9.0

1 0

[alsa-devel] [PATCH] ALSA: Move EXPORT_SYMBOL() in appropriate places
by Takashi Iwai 27 Feb '14

27 Feb '14

Just a cleanup to follow the standard coding style. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/hwdep.c | 3 +-- sound/core/rawmidi.c | 35 +++++++++++++++++------------------ 2 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index d6eb3ef3e3c7..69459e5f712e 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -395,6 +395,7 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, *rhwdep = hwdep; return 0; } +EXPORT_SYMBOL(snd_hwdep_new); static int snd_hwdep_free(struct snd_hwdep *hwdep) { @@ -567,5 +568,3 @@ static void __exit alsa_hwdep_exit(void) module_init(alsa_hwdep_init) module_exit(alsa_hwdep_exit) - -EXPORT_SYMBOL(snd_hwdep_new); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 801c86133937..6fc71a4c8a51 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -172,6 +172,7 @@ int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return 0; } +EXPORT_SYMBOL(snd_rawmidi_drop_output); int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) { @@ -203,6 +204,7 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) } return err; } +EXPORT_SYMBOL(snd_rawmidi_drain_output); int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) { @@ -217,6 +219,7 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return 0; } +EXPORT_SYMBOL(snd_rawmidi_drain_input); /* look for an available substream for the given stream direction; * if a specific subdevice is given, try to assign it @@ -354,6 +357,7 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, module_put(rmidi->card->module); return err; } +EXPORT_SYMBOL(snd_rawmidi_kernel_open); static int snd_rawmidi_open(struct inode *inode, struct file *file) { @@ -532,6 +536,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) module_put(rmidi->card->module); return 0; } +EXPORT_SYMBOL(snd_rawmidi_kernel_release); static int snd_rawmidi_release(struct inode *inode, struct file *file) { @@ -608,6 +613,7 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } +EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, struct snd_rawmidi_info __user *_info) @@ -655,6 +661,7 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, substream->active_sensing = !params->no_active_sensing; return 0; } +EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params * params) @@ -680,6 +687,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, runtime->avail_min = params->avail_min; return 0; } +EXPORT_SYMBOL(snd_rawmidi_input_params); static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, struct snd_rawmidi_status * status) @@ -935,6 +943,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_receive); static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, unsigned char __user *userbuf, @@ -977,6 +986,7 @@ long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, snd_rawmidi_input_trigger(substream, 1); return snd_rawmidi_kernel_read1(substream, NULL/*userbuf*/, buf, count); } +EXPORT_SYMBOL(snd_rawmidi_kernel_read); static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t count, loff_t *offset) @@ -1052,6 +1062,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_transmit_empty); /** * snd_rawmidi_transmit_peek - copy data from the internal buffer @@ -1108,6 +1119,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore(&runtime->lock, flags); return result; } +EXPORT_SYMBOL(snd_rawmidi_transmit_peek); /** * snd_rawmidi_transmit_ack - acknowledge the transmission @@ -1143,6 +1155,7 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) spin_unlock_irqrestore(&runtime->lock, flags); return count; } +EXPORT_SYMBOL(snd_rawmidi_transmit_ack); /** * snd_rawmidi_transmit - copy from the buffer to the device @@ -1164,6 +1177,7 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, return count; return snd_rawmidi_transmit_ack(substream, count); } +EXPORT_SYMBOL(snd_rawmidi_transmit); static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, const unsigned char __user *userbuf, @@ -1225,6 +1239,7 @@ long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream, { return snd_rawmidi_kernel_write1(substream, NULL, buf, count); } +EXPORT_SYMBOL(snd_rawmidi_kernel_write); static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, size_t count, loff_t *offset) @@ -1504,6 +1519,7 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, *rrawmidi = rmidi; return 0; } +EXPORT_SYMBOL(snd_rawmidi_new); static void snd_rawmidi_free_substreams(struct snd_rawmidi_str *stream) { @@ -1687,6 +1703,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, list_for_each_entry(substream, &rmidi->streams[stream].substreams, list) substream->ops = ops; } +EXPORT_SYMBOL(snd_rawmidi_set_ops); /* * ENTRY functions @@ -1725,21 +1742,3 @@ static void __exit alsa_rawmidi_exit(void) module_init(alsa_rawmidi_init) module_exit(alsa_rawmidi_exit) - -EXPORT_SYMBOL(snd_rawmidi_output_params); -EXPORT_SYMBOL(snd_rawmidi_input_params); -EXPORT_SYMBOL(snd_rawmidi_drop_output); -EXPORT_SYMBOL(snd_rawmidi_drain_output); -EXPORT_SYMBOL(snd_rawmidi_drain_input); -EXPORT_SYMBOL(snd_rawmidi_receive); -EXPORT_SYMBOL(snd_rawmidi_transmit_empty); -EXPORT_SYMBOL(snd_rawmidi_transmit_peek); -EXPORT_SYMBOL(snd_rawmidi_transmit_ack); -EXPORT_SYMBOL(snd_rawmidi_transmit); -EXPORT_SYMBOL(snd_rawmidi_new); -EXPORT_SYMBOL(snd_rawmidi_set_ops); -EXPORT_SYMBOL(snd_rawmidi_info_select); -EXPORT_SYMBOL(snd_rawmidi_kernel_open); -EXPORT_SYMBOL(snd_rawmidi_kernel_release); -EXPORT_SYMBOL(snd_rawmidi_kernel_read); -EXPORT_SYMBOL(snd_rawmidi_kernel_write); -- 1.9.0

1 0

[alsa-devel] [GIT PUL] ASoC updates for v3.14
by Mark Brown 27 Feb '14

27 Feb '14

The following changes since commit 57374bb491facf716c172126b7e9b99c9e3f8b45: Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus (2014-02-23 12:20:34 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc4-2 for you to fetch changes up to 23308e88201c2c0036da23450f7fed6fd64ae1ef: Merge remote-tracking branch 'asoc/fix/wm8958' into asoc-linus (2014-02-27 20:26:10 +0900) ---------------------------------------------------------------- ASoC: Updates for v3.14 A few more driver specific bug fixes, all driver specific things that only affect users of those devices. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: wm8958-dsp: Fix firmware block loading Mark Brown (5): ASoC: da732x: Mark DC offset control registers volatile Merge tag 'asoc-v3.14-rc3' into asoc-linus Merge tag 'asoc-v3.14-rc4' into asoc-linus Merge remote-tracking branches 'asoc/fix/da732x' and 'asoc/fix/sta32x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8958' into asoc-linus Takashi Iwai (1): ASoC: sta32x: Fix wrong enum for limiter2 release rate sound/soc/codecs/da732x.c | 12 ++++++++++++ sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 2 +- 3 files changed, 14 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: io: Clean up snd_soc_codec_set_cache_io()
by Xiubo Li 27 Feb '14

27 Feb '14

Now that all users have been converted to regmap and the config.reg_bits and config.val_bits can be setted by each user through regmap core API. So these two params are redundant here. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- @Mark and All I'm also thinking could we just discard snd_soc_codec_set_cache_io() calling from each individual driver to simply the code? And just bind it to devm_regmap_init_i2c() and devm_regmap_init_spi()... Is there any other limitations for snd_soc_codec_set_cache_io() usage? include/sound/soc.h | 1 - sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.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/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-io.c | 9 --------- 85 files changed, 83 insertions(+), 93 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ecfb334..0a1d732 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -404,7 +404,6 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, int snd_soc_codec_writable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, enum snd_soc_control_type control); int snd_soc_cache_sync(struct snd_soc_codec *codec); int snd_soc_cache_init(struct snd_soc_codec *codec); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 9381a76..052495f 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -325,7 +325,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ad193x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 5765c22..69b434c 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1379,7 +1379,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 9eb53c8..0066feb 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -804,7 +804,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 684fe91..9d4f878 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -392,7 +392,7 @@ static int ak4535_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4535->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 684b56f..5d81cbd 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -521,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f646c6..7e7621a 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -467,7 +467,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index deb2b44..46fd465 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -615,7 +615,7 @@ static int ak4671_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index ed50625..030dcda 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -905,7 +905,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) int ret; codec->control_data = alc5623->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d885056..2cd5d27 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1065,7 +1065,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) codec->control_data = alc5632->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 43737a27..d4f8714 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -140,7 +140,7 @@ static int cq93vc_probe(struct snd_soc_codec *codec) davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc->regmap; - snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 83c835d..988a00a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -509,7 +509,7 @@ static int cs4270_probe(struct snd_soc_codec *codec) /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 5caf75b..42564fb 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -488,7 +488,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) { int ret, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be455ea..d7a5967 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1112,7 +1112,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec) int ret; codec->control_data = cs42l52->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 06f4291..efa3b20 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1350,7 +1350,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) codec->control_data = cs42l73->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 01e55fc..3cfbab6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1076,7 +1076,7 @@ static int da7210_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); codec->control_data = da7210->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 439d103..bf6b840 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1398,7 +1398,7 @@ static int da7213_probe(struct snd_soc_codec *codec) struct da7213_platform_data *pdata = da7213->pdata; codec->control_data = da7213->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 05f07fb..f32f219 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1508,7 +1508,7 @@ static int da732x_probe(struct snd_soc_codec *codec) codec->control_data = da732x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to register codec.\n"); goto err; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index f118daa..187d8ce 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1387,7 +1387,7 @@ static int da9055_probe(struct snd_soc_codec *codec) struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); codec->control_data = da9055->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index cb736dd..54ea3cf 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1096,7 +1096,7 @@ static int isabelle_probe(struct snd_soc_codec *codec) codec->control_data = dev_get_regmap(codec->dev, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e533594..8061103 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1414,7 +1414,7 @@ static int lm49453_probe(struct snd_soc_codec *codec) codec->control_data = lm49453->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 31f9156..896ae28 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -136,7 +136,7 @@ static int max9768_probe(struct snd_soc_codec *codec) int ret; codec->control_data = max9768->regmap; - ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret) return ret; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4b9274d..5355e1a 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1914,7 +1914,7 @@ static int max98088_probe(struct snd_soc_codec *codec) regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index a75752f..d01ac5d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2220,7 +2220,7 @@ static int max98090_probe(struct snd_soc_codec *codec) codec->control_data = max98090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 4c22582..02666ce 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2238,7 +2238,7 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 82757eb..b7951b3 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -314,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index c605036..16725e9 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -615,7 +615,7 @@ static int mc13783_probe(struct snd_soc_codec *codec) int ret; codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 577fb87..3202eee 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -590,7 +590,7 @@ static int ml26124_probe(struct snd_soc_codec *codec) struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index ce199d3..3287fe7 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1574,7 +1574,7 @@ static int rt5631_probe(struct snd_soc_codec *codec) codec->control_data = rt5631->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1a1e115..26b9988 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1941,7 +1941,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640->codec = codec; codec->control_data = rt5640->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ab4754a..8758c3b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1354,7 +1354,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* setup i2c data ops */ codec->control_data = sgtl5000->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index bca7d02..671c45b 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -825,7 +825,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 806f3d8..ce4f890 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -652,7 +652,7 @@ static int ssm2518_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ssm2518->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 1294709..912fa5d 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -563,7 +563,7 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ssm2602->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ea78c17..93e67c8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -876,7 +876,7 @@ static int sta32x_probe(struct snd_soc_codec *codec) * then do the I2C transactions itself. */ codec->control_data = sta32x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); goto err; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index f15b0e3..9530e079 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -326,7 +326,7 @@ static int sta529_probe(struct snd_soc_codec *codec) int ret; codec->control_data = sta529->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 139f11f..70cb324 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -561,7 +561,7 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index ff5f23d..41186ef 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -296,7 +296,7 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); aic26->codec = codec; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6bd7e7..d79a9ec 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -614,7 +614,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb..598625b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1344,7 +1344,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8ae5027..a8dde7b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -786,7 +786,7 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 1e0a083..58b9fb0 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1557,7 +1557,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) codec->control_data = wm2200->regmap; codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index d3fa65f..5874238 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2345,7 +2345,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) wm5100->codec = codec; codec->control_data = wm5100->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffc..50593bb 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1762,7 +1762,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) codec->control_data = priv->core.arizona->regmap; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf1..9cb4257 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1590,7 +1590,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) codec->control_data = priv->core.arizona->regmap; priv->core.arizona->dapm = &codec->dapm; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index a183dcf..35c30a9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1507,7 +1507,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) codec->control_data = wm8350->regmap; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 6d684d9..10b1172 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1319,7 +1319,7 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) codec->control_data = wm8400->regmap; priv->codec = codec; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 7df7d45..e531b9d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -591,7 +591,7 @@ static int wm8510_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5dfd571..f5c17a0 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -398,7 +398,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989a..249961b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -869,7 +869,7 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948..3673b90 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -367,7 +367,7 @@ static int wm8711_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index cd89033..c2d2835 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -230,7 +230,7 @@ static int wm8728_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index d9655f9..3f877d2 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -584,7 +584,7 @@ static int wm8731_probe(struct snd_soc_codec *codec) int ret = 0, i; codec->control_data = wm8731->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index ecc4e87..8f14ed8 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -570,7 +570,7 @@ static int wm8737_probe(struct snd_soc_codec *codec) struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index dd02ebf..6a40489 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -429,7 +429,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_get; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); goto err_enable; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 78616a6..886d618 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -702,7 +702,7 @@ static int wm8750_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da9..f5e4d1a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1472,7 +1472,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); codec->control_data = wm8753->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 5bce210..ff2e087 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,7 +580,7 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ef82467..70276d7 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -430,7 +430,7 @@ static int wm8776_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 72d12bb..ae7f031 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -548,7 +548,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) codec->control_data = wm8804->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 43c2201..651908f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1180,7 +1180,7 @@ static int wm8900_probe(struct snd_soc_codec *codec) { int ret = 0, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b82b70a..0a92e99 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1902,7 +1902,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) wm8903->codec = codec; codec->control_data = wm8903->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 27299cd..f524481 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2064,7 +2064,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 87f032d..f53033b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -712,7 +712,7 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index d4dcaec..5762e3c 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -897,7 +897,7 @@ static int wm8955_probe(struct snd_soc_codec *codec) codec->control_data = wm8955->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f156010..1b90049 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -976,7 +976,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index ce8fa6e..f18950c 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -839,7 +839,7 @@ static int wm8961_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 62af9dc..4199e0e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3426,7 +3426,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962->codec = codec; codec->control_data = wm8962->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 67aba78..7dfee38 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -648,7 +648,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 6e16c43..3aa5821 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -593,7 +593,7 @@ static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a9e2f46..4564695 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -983,7 +983,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) */ wm8978->sysclk = WM8978_PLL; codec->control_data = wm8978->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 58f0551..ff0f005 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -995,7 +995,7 @@ static int wm8983_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index d786f2b..677253c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -997,7 +997,7 @@ static int wm8985_probe(struct snd_soc_codec *codec) wm8985 = snd_soc_codec_get_drvdata(codec); codec->control_data = wm8985->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 0277a76..88fec3c 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -814,7 +814,7 @@ static int wm8988_probe(struct snd_soc_codec *codec) int ret = 0; codec->control_data = wm8988->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 33f53ab..8acfa9a 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1291,7 +1291,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 32d2195..d288976 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1250,7 +1250,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 7b0630a..abc78d2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1494,7 +1494,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.series_startup = 1; codec->control_data = wm8993->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 699b527..cef8fdc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4001,7 +4001,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.codec = codec; codec->control_data = control->regmap; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 33ff361..f50b937 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2043,7 +2043,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995->codec = codec; codec->control_data = wm8995->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c8244af..14f27f25 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2635,7 +2635,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) codec->control_data = wm8996->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); goto err; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442c..4a36d1a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1055,7 +1055,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) codec->control_data = priv->core.arizona->regmap; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 721cee7..d4b8a0c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1264,7 +1264,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) codec->control_data = wm9081->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a07fe16..19285b0 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -526,7 +526,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) int ret; codec->control_data = wm9090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5a535b..86c5733 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1152,7 +1152,7 @@ static int soc_probe_codec(struct snd_soc_card *card, /* If the driver didn't set I/O up try regmap */ if (!codec->write && dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, SND_SOC_REGMAP); if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index add99e2..4b4f3d9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -69,8 +69,6 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * snd_soc_codec_set_cache_io: Set up standard I/O functions. * * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. * @control: Control bus used. * * Register formats are frequently shared between many I2C and SPI @@ -85,19 +83,13 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * volatile registers. */ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, enum snd_soc_control_type control) { - struct regmap_config config; int ret; - memset(&config, 0, sizeof(config)); codec->write = hw_write; codec->read = hw_read; - config.reg_bits = addr_bits; - config.val_bits = data_bits; - switch (control) { case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ @@ -123,7 +115,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, enum snd_soc_control_type control) { return -ENOTSUPP; -- 1.8.4

3 2

[alsa-devel] [PATCH] ASoC: sta32x: Fix wrong enum for limiter2 release rate
by Takashi Iwai 27 Feb '14

27 Feb '14

There is a typo in the Limiter2 Release Rate control, a wrong enum for Limiter1 is assigned. It must point to Limiter2. Spotted by a compile warning: In file included from sound/soc/codecs/sta32x.c:34:0: sound/soc/codecs/sta32x.c:223:29: warning: ‘sta32x_limiter2_release_rate_enum’ defined but not used [-Wunused-variable] static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ include/sound/soc.h:275:18: note: in definition of macro ‘SOC_ENUM_DOUBLE_DECL’ struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ sound/soc/codecs/sta32x.c:223:8: note: in expansion of macro ‘SOC_ENUM_SINGLE_DECL’ static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, ^ Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Mark, this comes up after merging your pull request. Interestingly, the warning is triggered because const was temporarily reduced along my changes. Since my whole patch series added the const in the macro definition, I didn't see this. Ditto for linux-next, where no relevant warning is seen. I'd like to put this as a quick fix for the next 3.14 pull request. So, if you don't mind, I'm going to commit to my tree for-linus branch. Takashi sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ea78c172538c..2735361a4c3c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both -- 1.9.0

2 3

[alsa-devel] Unsupported Valley View 'LPE' adapter?
by Adam Williamson 27 Feb '14

27 Feb '14

Hi, folks. I recently bought a Dell Venue 8 Pro - a Bay Trail / Valley View-based Intel tablet - to try and get Fedora running on it. Among the many things I'm currently fighting :) , the sound doesn't work. snd-hda-intel is not loaded at all, and there are no audio devices present at any level. In Windows' Device Manager I see an 'Intel SST Audio Device' with PCI ID 0x86:0f28 . Indeed, the PCI ID database has this listed as "ValleyView LPE Audio Controller" - http://pci-ids.ucw.cz/read/PC/8086/0f28 . But hda-intel does not have 0f28 as a supported device. The commit for 'Bay Trail' audio support added the ID 0f04: https://lkml.org/lkml/2013/5/16/38 So...what is this 0f28? Is support for it going to be added? Is there any info I can provide to help? (I can boot Linux on the tablet and get info out of it _just barely_, but it's pretty painful - there's no wifi and an install to 'disk' (really a usb stick) doesn't work too well, so I'm stuck working in a live image with no network). -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | XMPP: adamw AT happyassassin . net http://www.happyassassin.net

6 13

[alsa-devel] [PATCH v4 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 27 Feb '14

27 Feb '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v4: 1. Use the generic dmaengine pcm codes instead of sirf pcm driver. 2. Add TLV information into internal audio codec driver. 3. Use the MMIO regmap instead of custom read and write callbacks. 4. Remove spin_lock. 5. Change some variable names. 6. The SiRF internal audio driver split into codec, port. 7. Add binding documents. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode ASoC: sirf: Add bt-sco card .../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/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sirf-audio-codec.c | 585 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + 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/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 438 +++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 +++ sound/soc/sirf/sirf-usp.c | 448 +++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 ++++++++++ 24 files changed, 2745 insertions(+), 0 deletions(-) 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/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt 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/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 create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

3 10

[alsa-devel] [PATCH] ASoC: fsl-sai: Add SND_SOC_DAIFMT_DSP_A/B support.
by Xiubo Li 27 Feb '14

27 Feb '14

o Add SND_SOC_DAIFMT_DSP_A support. o Add SND_SOC_DAIFMT_DSP_B support. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 26d9f5e..c4a4231 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -108,15 +108,44 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Data on rising edge of bclk, frame low, 1clk before data */ + /* + * Frame low, 1clk before data, one word length for frame sync, + * frame sync starts one serial clock cycle earlier, + * that is, together with the last bit of the previous + * data word. + */ val_cr2 &= ~FSL_SAI_CR2_BCP; val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_LEFT_J: - /* Data on rising edge of bclk, frame high, 0clk before data */ + /* + * 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_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; + case SND_SOC_DAIFMT_DSP_A: + /* + * Frame high, 1clk before data, one bit for frame sync, + * frame sync starts one serial clock cycle earlier, + * that is, together with the last bit of the previous + * 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; + case SND_SOC_DAIFMT_DSP_B: + /* + * Frame high, one bit for frame sync, + * 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: /* To be done */ default: @@ -219,7 +248,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; - val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + if (!sai->is_dsp_mode) + val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); @@ -245,6 +276,10 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); u32 tcsr, rcsr; + /* + * The transmitter bit clock and frame sync are to be + * used by both the transmitter and receiver. + */ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, ~FSL_SAI_CR2_SYNC); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, @@ -261,6 +296,10 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr &= ~FSL_SAI_CSR_FRDE; } + /* + * It is recommended that the transmitter is the last enabled + * and the first disabled. + */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 1571459..e432260 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -103,6 +103,7 @@ struct fsl_sai { bool big_endian_regs; bool big_endian_data; + bool is_dsp_mode; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; -- 1.8.4

2 1

[alsa-devel] [PATCH] Asoc: codecs: Remove tlv320aic32x4 from compatibles of tlv320aic3x
by Markus Pargmann 27 Feb '14

27 Feb '14

This reverts tlv320aic32x4 as compatible for tlv320aic3x as it has its own bindings now. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 9d8ea14..5e6040c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -6,7 +6,6 @@ Required properties: - compatible - "string" - One of: "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic32x4" - TLV320AIC32x4 "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 "ti,tlv320aic3106" - TLV320AIC3106 -- 1.8.5.3

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 27 Feb '14

27 Feb '14

The following changes since commit 895be5b31e5175bef575008aadb4f0a27b850daa: Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus (2014-02-19 13:13:58 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc4 for you to fetch changes up to 57374bb491facf716c172126b7e9b99c9e3f8b45: Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus (2014-02-23 12:20:34 +0900) Getting a bit large but they're all actual bug fixes unfortunately. ---------------------------------------------------------------- ASoC: Fixes for v3.14 A somewhat large set of fixes here due to the identification of some systematic problems with hard to use APIs in the subsystem. Takashi did a lot of work to address the enumeration API which uncovered a number of off by one bugs caused by confusing APIs while Charles addressed issues in the locking around DAPM. ---------------------------------------------------------------- Charles Keepax (3): ASoC: dapm: Correct regulator bypass error messages Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions Lars-Peter Clausen (1): ASoC: sta32x: Fix cache sync Mark Brown (4): Merge tag 'asoc-v3.14-rc3' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/ad1980' and 'asoc/fix/isabelle' into asoc-linus Merge remote-tracking branches 'asoc/fix/sta32x', 'asoc/fix/wm8400', 'asoc/fix/wm8770', 'asoc/fix/wm8900' and 'asoc/fix/wm8994' into asoc-linus Takashi Iwai (7): ASoC: sta32x: Fix array access overflow ASoC: wm8770: Fix wrong number of enum items ASoC: wm8900: Fix the wrong number of enum items ASoC: wm8994: Fix the wrong number of enum items ASoC: ad1980: Fix wrong number of items for capture source ASoC: isabelle: Fix the wrong number of items in enum ctls ASoC: wm8400: Fix the wrong number of enum items drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ----- include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/isabelle.c | 52 +++++++------ sound/soc/codecs/sta32x.c | 74 +++++++++---------- sound/soc/codecs/wm8400.c | 34 +++++---- sound/soc/codecs/wm8770.c | 4 +- sound/soc/codecs/wm8900.c | 44 +++++------ sound/soc/codecs/wm8994.c | 135 ++++++++++++++++++---------------- sound/soc/soc-dapm.c | 139 +++++++++++++++++++++++++++++++---- 11 files changed, 317 insertions(+), 208 deletions(-)

1 0

[alsa-devel] tlv320aic3x regcache issue
by Jack Mitchell 26 Feb '14

26 Feb '14

Hi, I have come across an issue where the regcache mechanism used in the tlv320aic3x driver is causing problems. When switching from 44.1khz to 48khz some of the registers are not written due to the default register cache holding the same value as being written. However, the codec requires these registers to be written in order to trigger the change, this has been proven by changing the register default to something different, therefore forcing a write at the correct time and it triggers the change as expected. What would be the best way to force these registers to be written? Is there some sort of force write flag, or a way to turn off the cache just during this portion of the configuration? Any direction in the proper way of doing this would be much appreciated. Cheers, Jack -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer Cambridgeshire, UK http://www.embed.me.uk --

1 0

[alsa-devel] tlv320aic3x regcache issue
by Jack Mitchell 26 Feb '14

26 Feb '14

Hi, I have come across an issue where the regcache mechanism used in the tlv320aic3x driver is causing problems. When switching from 44.1khz to 48khz some of the registers are not written due to the default register cache holding the same value as being written. However, the codec requires these registers to be written in order to trigger the change, this has been proven by changing the register default to something different, therefore forcing a write at the correct time and it triggers the change as expected. What would be the best way to force these registers to be written? Is there some sort of force write flag, or a way to turn off the cache just during this portion of the configuration? Any direction in the proper way of doing this would be much appreciated. Cheers, Jack -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer Cambridgeshire, UK http://www.embed.me.uk --

1 0

[alsa-devel] [PATCH 00/48] Rewrite kernel messages with standard printk helpers
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is another long series of patches for trivial cleanups. This time, they replace the printk and snd_print*() calls in ALSA PCI and USB drivers with dev_err() and co. This will give nicer and more consistent outputs. The only drawback by these replacements is that alsa-info.sh won't be able to catch the dmesg lines because it doesn't contain any longer the word "ALSA", but only show the module name like "snd_intel8x0". I'm going to fix alsa-info.sh on git soon later, but if you guys think "oh no it's disaster!", let me know. We need to reconsider a better way to put the known markers (like explicitly putting "ALSA" at the beginning.) Takashi

1 49

[alsa-devel] [PATCH 2/2] ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI
by Alexander Shiyan 26 Feb '14

26 Feb '14

This patch adds the devicetree documentation for the Cirrus Logic CLPS711X Digital Audio Interface (DAI). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- .../bindings/sound/cirrus,clps711x-dai.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt new file mode 100644 index 0000000..718599e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt @@ -0,0 +1,20 @@ +* Cirrus Logic CLPS711X Digital Audio Interface (DAI) + +Required properties: +- compatible: Should be "cirrus,clps711x-dai". +- reg: Should contain registers location and length. +- clocks: A clock specifier for the CPU PLL clock. +- clock-names: The CPU PLL clock should be named "pll". +- interrupts: Should contain DAI interrupt number. +- syscon: phandle to syscon device node which control DAI (SYSCON3). + +Example: + dai: dai@80002000 { + #sound-dai-cells = <0>; + compatible = "cirrus,clps711x-dai"; + reg = <0x80002000 0x604>; + clocks = <&clks 3>; + clock-names = "pll"; + interrupts = <32>; + syscon = <&syscon3>; + }; -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ALSA: rme96: Convert to the new pm_ops
by Takashi Iwai 26 Feb '14

26 Feb '14

This driver slipped from the last rewrite. Just convert to the new standard pm ops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme96.c | 34 ++++++++++++++++++---------------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 00d46116e04a..f5014a8e69ad 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -240,7 +240,7 @@ struct rme96 { u8 rev; /* card revision number */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 playback_pointer; u32 capture_pointer; void *playback_suspend_buffer; @@ -1570,7 +1570,7 @@ snd_rme96_free(void *private_data) pci_release_regions(rme96->pci); rme96->port = 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP vfree(rme96->playback_suspend_buffer); vfree(rme96->capture_suspend_buffer); #endif @@ -2372,13 +2372,12 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP -static int -snd_rme96_suspend(struct pci_dev *pci, - pm_message_t state) +static int rme96_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -2407,10 +2406,10 @@ snd_rme96_suspend(struct pci_dev *pci, return 0; } -static int -snd_rme96_resume(struct pci_dev *pci) +static int rme96_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct rme96 *rme96 = card->private_data; pci_restore_state(pci); @@ -2451,7 +2450,11 @@ snd_rme96_resume(struct pci_dev *pci) return 0; } -#endif +static SIMPLE_DEV_PM_OPS(rme96_pm, rme96_suspend, rme96_resume); +#define RME96_PM_OPS &rme96_pm +#else +#define RME96_PM_OPS NULL +#endif /* CONFIG_PM_SLEEP */ static void snd_rme96_card_free(struct snd_card *card) { @@ -2488,7 +2491,7 @@ snd_rme96_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); if (!rme96->playback_suspend_buffer) { snd_printk(KERN_ERR @@ -2547,10 +2550,9 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = snd_rme96_remove, -#ifdef CONFIG_PM - .suspend = snd_rme96_suspend, - .resume = snd_rme96_resume, -#endif + .driver = { + .pm = RME96_PM_OPS, + }, }; module_pci_driver(rme96_driver); -- 1.9.0

1 0

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow to specify a client name as parameters
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Hi Takashi, I reworked the patches and hope to be the final ones. J You will find attached at each of the emails. Kind Regards, Valentin Corfu Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

2 1

[alsa-devel] [PATCH - speaker-test 1/1] speaker-test: add --force-frequency option to allow hz outside range
by erik＠caspar.alsa-project.org 26 Feb '14

26 Feb '14

From: Erik Ackermann <domohawk(a)gmail.com> Signed-off-by: Erik Ackermann <domohawk(a)gmail.com> diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 261f399..22927f2 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -106,6 +106,7 @@ static snd_pcm_uframes_t period_size; static const char *given_test_wav_file = NULL; static char *wav_file_dir = SOUNDSDIR; static int debug = 0; +static int force_frequency = 0; static int in_aborting = 0; static snd_pcm_t *pcm_handle = NULL; @@ -1015,6 +1016,7 @@ static void help(void) "-w,--wavfile Use the given WAV file as a test sound\n" "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" + "-X,--force-frequency force frequencies outside the 30-5000hz range\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1057,6 +1059,7 @@ int main(int argc, char *argv[]) { {"wavfile", 1, NULL, 'w'}, {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, + {"force-frequncy", 1, NULL, 'X'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1078,7 +1081,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1114,8 +1117,6 @@ int main(int argc, char *argv[]) { break; case 'f': freq = atof(optarg); - freq = freq < 30.0 ? 30.0 : freq; - freq = freq > 5000.0 ? 5000.0 : freq; break; case 'b': buffer_time = atoi(optarg); @@ -1173,6 +1174,9 @@ int main(int argc, char *argv[]) { case 'd': debug = 1; break; + case 'X': + force_frequency = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': chmap = optarg; @@ -1190,6 +1194,11 @@ int main(int argc, char *argv[]) { exit(EXIT_SUCCESS); } + if (!force_frequency) { + freq = freq < 30.0 ? 30.0 : freq; + freq = freq > 5000.0 ? 5000.0 : freq; + } + if (test_type == TEST_WAV) format = SND_PCM_FORMAT_S16_LE; /* fixed format */ -- 1.8.3.2

4 7

[alsa-devel] MAYA44 USB audio device no inputs control (again)
by Leo Combes 26 Feb '14

26 Feb '14

Hi, I have bought a Maya 44 USB, with 4 output and 4 input channels. http://www.esi-audio.com/products/maya44usb/ The soundcard works fine in playback, but in recording has some problems, specificly there is no way to control the volume of the recording on inputs. It seems to be default setted to 50%. Alsamixer, in playback section shows 3 controls, "speaker", "line" and "line1". But, if I choose capture, it shows "This sound device does not have any capture controls". It seems like there are controls, and they work fine, running on Windows. I searched the problem on Google and I found a lot of users with the same problem. The most interesting thread that I found, is one on this mailist, sadly after some messages, user didn't catch up with the conversation: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-November/046242.h… I have a hardware mixer connected to 2 inputs of the Maya. If I try to recording with Audacity or any other software, the volume of recording is never greater than -6dB. If I try increase the volume in the mixer to get louder recording, the only thing I get is a completely distorted sound not louder than -6dB, what tells me that Maya inputs have a presetted volume at 50%. Besides, if I increase the volume in the mixer, Audicity's vumeter confirm that it never go further than -6dB. Following the thread that the user left unfinished, here goes the information that I get with my system, to see if could find a solution. leo-ThinkPad leo # aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=MID HDA Intel MID, CX20585 Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample mixing device dsnoop:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct sample snooping device hw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Direct hardware device without any conversions plughw:CARD=MID,DEV=0 HDA Intel MID, CX20585 Analog Hardware device with all software conversions hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 HDMI Audio Output dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample mixing device dmix:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample mixing device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct sample snooping device dsnoop:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct sample snooping device hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Direct hardware device without any conversions hw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=7 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=8 HDA NVidia, HDMI 0 Hardware device with all software conversions plughw:CARD=NVidia,DEV=9 HDA NVidia, HDMI 0 Hardware device with all software conversions sysdefault:CARD=USB MAYA44 USB+, USB Audio Default Audio Device front:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Front speakers surround40:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=USB,DEV=0 MAYA44 USB+, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=USB,DEV=0 MAYA44 USB+, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample mixing device dsnoop:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct sample snooping device hw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Direct hardware device without any conversions plughw:CARD=USB,DEV=0 MAYA44 USB+, USB Audio Hardware device with all software conversions leo-ThinkPad leo # lsmod | grep snd snd_hda_codec_hdmi 41684 4 snd_usb_audio 155972 3 snd_hda_codec_conexant 57181 1 snd_usbmidi_lib 29576 1 snd_usb_audio snd_hda_intel 57183 2 snd_hda_codec 194881 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel snd_hwdep 13613 2 snd_usb_audio,snd_hda_codec snd_pcm 107140 6 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18798 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30465 2 snd_usbmidi_lib,snd_seq_midi snd_seq 66061 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 30038 2 snd_pcm,snd_seq snd 73802 22 snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,thinkpad_acpi,snd_seq_device,snd_seq_midi soundcore 12680 1 snd leo-ThinkPad leo # amixer contents numid=2,iface=MIXER,name='Line Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Line Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=3,iface=MIXER,name='Line Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=5,iface=MIXER,name='Line Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=0.00dB,max=0.99dB numid=6,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=255,255 | dBminmax-min=0.00dB,max=0.99dB numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=4,min=0,max=36,step=0 : values=3,4,0,0 | | TLV size error (257, 8, 0)! leo-ThinkPad leo # amixer -c2 Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.99dB] [on] Front Right: Playback 255 [100%] [0.99dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Simple mixer control 'Line',1 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [0.00dB] [off] Thanks in advance!

1 0

[alsa-devel] [PATCH 1/1] ALSA: compress: no need to set metadata before next track
by Richard Fitzgerald 26 Feb '14

26 Feb '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7a20897..5a351c7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -744,12 +744,6 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; - /* you can signal next track isf this is intended to be a gapless stream - * and current track metadata is set - */ - if (stream->metadata_set == false) - return -EPERM; - retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); if (retval != 0) return retval; -- 1.7.2.5

1 0

[alsa-devel] [PATCH] ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio'
by Jarkko Nikula 26 Feb '14

26 Feb '14

Some kernel configurations can cause following build error: sound/soc/intel/sst-baytrail-ipc.c: In function ‘sst_byt_get_dsp_position’: sound/soc/intel/sst-baytrail-ipc.c:744:2: error: implicit declaration of function ‘memcpy_fromio’ [-Werror=implicit-function-declaration] memcpy_fromio(&fw_tstamp, ^ cc1: some warnings being treated as errors Fix this by including <asm/io.h> explicitly. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Also sst-dsp., sst-baytrail-dsp.c and sst-haswell-dsp.c are using memcpy_fromio/_toio but there <asm/io.h> seems to come via <linux/dma-mapping.h> or <linux/io.h> so I didn't touch them. --- sound/soc/intel/sst-baytrail-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 8c91a68b90bc..8b067e34094e 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -27,6 +27,7 @@ #include <linux/kthread.h> #include <linux/firmware.h> #include <asm/div64.h> +#include <asm/io.h> #include "sst-baytrail-ipc.h" #include "sst-dsp.h" -- 1.8.5.3

3 3

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow number of channels
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

1 0

[alsa-devel] [PATCH] Bugfix: Fix resampling when client and slave both use format float
by Maarten Baert 26 Feb '14

26 Feb '14

I found a bug in libasound. When both the client and the slave (in my case, the JACK plugin) try to use the float format, but the sample rate or channel count does not match, libasound *should* insert linear conversion plugins to convert from float to linear, then resample/remap channels, and then convert back to float (because apparently the resamplers and channel remapper don't support floating point, only 'linear' i.e. integers). Currently this doesn't work, snd_pcm_plug_change_format doesn't know what to do and simply returns EINVAL. As a result, snd_pcm_hw_params fails even though the HW params were perfectly valid (it indicates that both the float format and any sample rate are supported). In my test, this broke audio for WINE (and any other application that tries to use float, such as aplay with the right settings) when I wanted to use the JACK plugin (which only supports the float format). This patch fixes this bug by doing a conversion to S16 and back when resampling or remapping is needed. And while I was at it, I also removed a check that had no effect because the exact same check is already being done at the start of the function. I still think it's a bit silly that libasound requires integers for resampling, because both libsamplerate and libspeex use float internally for resampling. So now ALSA is literally doing a float-to-s16-to-float-to-s16-to-float conversion. But changing that would have been a lot harder. Maarten Baert

3 10

[alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices
by Takashi Sakamoto 26 Feb '14

26 Feb '14

This series of patch is to update patches which I've posted below. [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070424.h… [RFC][PATCH v2 00/13] speakers: Add support for capture/playback of PCM/MIDI http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071071.ht… This series may be last RFC. Repository: https://github.com/takaswie/sound For backport to Linux kernel 3.11 or later: https://github.com/takaswie/snd-firewire-improve For maintainers: I have 4 items. 1.expected impacts to user-land Refer to my last message. 2. Address allocation for Fireworks drivers Related to this suggestion: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071120.ht… The alternative way is to implement: 1.'AV/C SIGNAL FORMAT' commanids defined in 'AV/C Digital Interface Command Set General Specification Version 4.2 (Sep 2004, 1394TA)' - 'snd-firewire-lib' has these commands 2.'AV/C DESCRIPTOR' mechanism defined in 'Enhancements to the AV/C General Specification 3.0 Version 1.1 (Oct 2000, 1394TA)' 3.'AV/C SIGNAL SOURCE' commands defined in 'AV/C Music Subunit 1.0' - I don't have this specification. They cost higher than actual effects. Especially, 2 is a bit complicated and need much codes. Actually FFADO has all of them. And snd-fireworks gives hwdep interface for application to command/response. There is currently several ways in user-land so I think it good to keep the address non-relocatable yet. 3.Support for devices with fatal quirk M-Audio 'Firewire 1814' and 'ProjectMix I/O' has BeBoB firmware which is heavily customized. See last patch. They have a time not to respond transaction. As long as I experienced, the time may be 300 msec per 1 sec. As a result, snd-bebob often fail to handle them. Especially, they can ignore transactions for resource management of connection/channel/bandwidth. This is a fatal for host system. But there is no user-land driver for these devices and I received some requests from FFADO users to support them. I hope to get your opinion about this support. 4.Rest of my work I've finished almost work for them. If I receive no big problems from developers and testers, I have a plan to post merge-request. Updates since previous patches: snd-firewire-lib: - remove wrong line to regain 'stream_running' member to 'struct fwspk' - allow drivers to indicate timeout value - remove 'amdtp_stream_midi_running' snd-bebob: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member fron 'struct snd_bebob' - update device entry snd-fireworks: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member from 'struct snd_fireworks' snd-firewire-speakers: - rename to snd-oxfw and obsolete snd-oxfw: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - fix lock mechanism of streaming by hwdep - fix wrong formation for Lacie FWSpeaker and Griffin Firewave - remove 'fw_unit_get()/fw_unit_put()' because helper function do this Confirm to work with: snd-fireworks: - Echo Audio AudioFire4 - Echo Audio AudioFire8 (by tester) - Echo Audio AudioFirePre8 - Echo Audio AudioFire12 snd-bebob: - Edirol FA-66 (by tester) - M-Audio Ozonic - M-Audio Firewire Solo - M-Audio Firewire Audiophile - M-Audio Firewire 410 - M-Audio Firewire 1814 (by tester) - PreSonus FIREBOX - PreSonus FP10 (former known as FIREPOD, by tester) - Yamaha GO44 - Yamaha GO46 snd-oxfw: - Behringer F-Control Audio 202 Takashi Sakamoto (52): firewire-lib: Rename functions, structure, member for AMDTP 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 sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping 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 some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface 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: 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: Send a cue to load firmware for M-Audio Firewire series speakers: Rename to oxfw oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add some AV/C commands to get stream formation and supported sample rate oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 5 +- include/uapi/sound/firewire.h | 23 +- sound/firewire/Kconfig | 76 +- sound/firewire/Makefile | 5 +- sound/firewire/amdtp.c | 905 ++++++++++++++++------ sound/firewire/amdtp.h | 161 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 453 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 178 +++++ sound/firewire/bebob/bebob_pcm.c | 419 ++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 873 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 186 +++++ sound/firewire/fireworks/fireworks_pcm.c | 443 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 356 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 264 +++++++ sound/firewire/oxfw/oxfw.h | 146 ++++ sound/firewire/oxfw/oxfw_command.c | 162 ++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++ sound/firewire/oxfw/oxfw_hwdep.c | 195 +++++ sound/firewire/oxfw/oxfw_midi.c | 176 +++++ sound/firewire/oxfw/oxfw_pcm.c | 399 ++++++++++ sound/firewire/oxfw/oxfw_proc.c | 60 ++ sound/firewire/oxfw/oxfw_stream.c | 582 ++++++++++++++ sound/firewire/speakers.c | 830 -------------------- 45 files changed, 10576 insertions(+), 1173 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 create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_control.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2 Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

4 62

[alsa-devel] [PATCH] ASoC: da732x: Mark DC offset control registers volatile
by Mark Brown 26 Feb '14

26 Feb '14

From: Mark Brown <broonie(a)linaro.org> The driver reads from the DC offset control registers during callibration but since the registers are marked as volatile and there is a register cache the values will not be read from the hardware after the first reading rendering the callibration ineffective. It appears that the driver was originally written for the ASoC level register I/O code but converted to regmap prior to merge and this issue was missed during the conversion as the framework level volatile register functionality was not being used. Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/da732x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 8e2c74d..798eb43 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, -- 1.9.0.rc3

2 7

[alsa-devel] Fwd: i.MX audio support
by jwoodward 26 Feb '14

26 Feb '14

-------- Original Message -------- Subject: i.MX audio support Date: Wed, 26 Feb 2014 09:33:06 +0000 From: jwoodward <jwoodward(a)tensor.co.uk> To: s.hauer(a)pengutronix.de Hi Sascha, I am having trouble implimenting the wm8731 codec with the i.MX21. I have read your article https://lwn.net/Articles/363406/ which does not fill me with confidence on this. It is at the stage of finding the i2c device, and setting up the transmit line for AC97, I am not very experienced with audio and would appreciate any advise possible? Here are a few traces I have added: [ 2.935589] wm8731_i2c_probe: i2c_set_clientdata [ 2.989318] imx_ssi_init [ 2.997084] imx_ssi_probe [ 3.007482] imx_ssi_probe [ 3.015846] setup_channel_to_ac97 [ 3.029115] imx-fiq-pcm-audio imx-fiq-pcm-audio.1: failed to claim fiq: -16 [ 3.049930] imx-fiq-pcm-audio: probe of imx-fiq-pcm-audio.1 failed with error -16 [ 3.075173] ALSA device list: [ 3.084401] No soundcards found. Attached is a schematic of our interface. Many thanks, Jon -- Regards, __| __| __| Jonathan Woodward __| __| __| Embedded Developer __| __| __| Tensor plc jwoodward(a)tensor.co.uk Tel: +44 (0)1480 215530 http://www.tensor.co.uk Fax: +44 (0)1480 223966 ---------------------------------------------------------------------------------- Important Email Information The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and unlawful. If you are not the intended addressee please contact the sender and dispose of this email. Any opinions expressed in this e-mail are those of the individual and not necessarily those of Tensor plc or any companies within the Tensor group. Any prices for the supply of goods or services are only valid if supported by a formal written quotation. This communication is for information only and shall not create or change any contractual relationship. Every effort has been made to ensure that any attachment to this mail does not contain a virus. Whilst we have taken every reasonable precaution to minimise this risk, neither it nor the sender can accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening the attachment. ---------------------------------------------------------------------------------- Tensor plc, Hail Weston House, Hail Weston, St. Neots, Cambs, PE19 5JY, UK Registered in England number 03670909 - VAT Number 761 546 815

1 0

[alsa-devel] I lost one SSID for HP mute led
by Kailang 26 Feb '14

26 Feb '14

Hi Takashi, I lost one SSID. I add it as attach patch. Thanks. BR, Kailang

2 1

[alsa-devel] alsa on non-linux
by Patrick Welche 26 Feb '14

26 Feb '14

I am currently successfully running alsa-lib 1.0.22 with oss plugins on a non-linux box. The advantage is that programs written to use libasound work. I just tried to update alsa-lib, and see that now alsa-lib directly includes headers like linux/types.h, and all protection, such as #if defined(LINUX) || defined(__LINUX__) || defined(__linux__) has been removed. This means that compilation on non-linux is essentially impossible. Is it that there is now a different way of obtaining that alsa lib front end / oss back end layer? Cheers, Patrick

3 14

[alsa-devel] [PATCH 00/10] A few more ALSA and HD-audio core updates
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is a series of patches to cleanup / enhance the ALSA core and HD-audio driver codes. A part of them are the patches I already submitted as RFC shortly ago (patch 4 & patch 5), another part are for better sysfs managment (part 1 & 2). The rest half are about HD-audio, for creating the device struct for each codec, followed by a cleanup patch to replace printk with dev_err() & co. Takashi

1 10

[alsa-devel] CA0132 6 Channel Output
by Sean McCully 26 Feb '14

26 Feb '14

I have one of these sound cards that came with my motherboard, /http://www.asrock.com/microsite/intelx79/GameBlaster.html/[1] Two Channel Output from one audio Jack works perfectly, but there seems to be a problem with using the additional audio jacks for 6 channel "surround sound". http://www.alsa-project.org/db/?f=a8ae0aa3c0da4d697c1df5f403001dff50bb6e51 pasuspender -- speaker-test --channels=6 --device=surround51:0 speaker-test 1.0.27.2 Playback device is surround51:0 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise Channels count (6) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument $ aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System pulse PulseAudio Sound Server sysdefault:CARD=PCH HDA Intel PCH, CA0132 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output -- -- Sean McCully -------- [1] http://www.asrock.com/microsite/intelx79/GameBlaster.html

2 1

[alsa-devel] [PATCH] ASoC: cirrus: Remove excess dependencies on SND_SOC
by Alexander Shiyan 26 Feb '14

26 Feb '14

Configuration for Cirrus Logic audio support is included only if SND_SOC symbol selected, so no reason to check it once more. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 06f938d..c872dac 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC + depends on ARCH_EP93XX || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: fsl-sai: fix Freescale SAI DAI format setting.
by Xiubo Li 26 Feb '14

26 Feb '14

o Fix some bugs of fsl_sai_set_dai_fmt_tr(). o Add SND_SOC_DAIFMT_LEFT_J support. o Add SND_SOC_DAIFMT_CBS_CFM support. o Add SND_SOC_DAIFMT_CBM_CFS support. o And SND_SOC_DAIFMT_RIGHT_J need to be done in the future. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++--------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa65af..26d9f5e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -105,35 +105,47 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, else val_cr4 |= FSL_SAI_CR4_MF; + /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - val_cr4 |= FSL_SAI_CR4_FSE; + /* Data on rising edge of bclk, frame low, 1clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high, 0clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; + case SND_SOC_DAIFMT_RIGHT_J: + /* To be done */ default: return -EINVAL; } + /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert both clocks */ + val_cr2 ^= FSL_SAI_CR2_BCP; + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_IB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert bit clock */ + val_cr2 ^= FSL_SAI_CR2_BCP; break; case SND_SOC_DAIFMT_NB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Invert frame clock */ + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_NB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Nothing to do for both normal cases */ break; default: return -EINVAL; } + /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; @@ -143,6 +155,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, 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; } -- 1.8.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names
by Jarkko Nikula 26 Feb '14

26 Feb '14

Use more sensible kcontrol names than "Int Mic" and "Ext Spk". Speakers especially are usually integrated devices in sales models. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 038547b48b2d..672fedbfa680 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -29,27 +29,27 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Int Mic"}, + {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOLP"}, - {"Ext Spk", NULL, "SPOLN"}, - {"Ext Spk", NULL, "SPORP"}, - {"Ext Spk", NULL, "SPORN"}, + {"Speaker", NULL, "SPOLP"}, + {"Speaker", NULL, "SPOLN"}, + {"Speaker", NULL, "SPORP"}, + {"Speaker", NULL, "SPORN"}, }; static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), }; static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, @@ -113,8 +113,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headphone"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_sync(dapm); return ret; -- 1.8.5.3

3 3

[alsa-devel] [PATCH] ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware
by Jarkko Nikula 26 Feb '14

26 Feb '14

I swear I tested missing firmware in commit e5161d7987f1 ("ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing"). Unfortunately same wasn't done in commit 6dda27cbbd1d ("ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform") which will cause NULL pointer dereference in sst_acpi_fw_cb() when printing the error since sst_acpi->mach is not set. Fix this obvious error by setting the sst_acpi->mach in sst_acpi_probe(). Reported-by: Mika Westerberg <mika.westerberg(a)linux.intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-acpi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 0bb43169e146..5d06eecb6198 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -139,6 +139,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; + sst_acpi->mach = mach; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; -- 1.8.5.3

4 3

[alsa-devel] [patch] ASoC: Intel: sst-firmware: missing curly braces (harmless)
by Dan Carpenter 26 Feb '14

26 Feb '14

There were some curly braces intended here, but the code actually works the same either way so it's not a bug. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index b6f9b5ecb66a..51b56a473bc5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -293,7 +293,7 @@ static int block_module_prepare(struct sst_module *module) /* enable each block so that's it'e ready for module P/S data */ list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->enable) + if (block->ops && block->ops->enable) { ret = block->ops->enable(block); if (ret < 0) { dev_err(module->dsp->dev, @@ -301,6 +301,7 @@ static int block_module_prepare(struct sst_module *module) block->type, block->index); goto err; } + } } return ret;

2 1

[alsa-devel] [patch 1/2] ASoC: sst-haswell-ipc: restore IRQs on error
by Dan Carpenter 26 Feb '14

26 Feb '14

This should be spin_unlock_irqrestore() instead of spin_unlock() Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 668d486520ae..552aebf2c8f5 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -527,7 +527,7 @@ static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, msg = msg_get_empty(hsw); if (msg == NULL) { - spin_unlock(&hsw->dsp->spinlock); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return -EBUSY; }

3 2

[alsa-devel] [PATCH v4] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 25 Feb '14

25 Feb '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- 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. 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. * Use regmap_read/write instead of rt286_read/write to read write codec. * Use the standard jack detection APIs for jack detection. * Power off on bias level STANDBY. * Add .symmetric_rates = 1. * Add gpio2_en and remove irq_en in platform data. * Other minor changes. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1369 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 159 ++++++ 5 files changed, 1553 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 f2383eb..889b69f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -67,6 +67,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -365,6 +366,9 @@ config SND_SOC_PCM512x tristate "Texas Instruments PCM512x CODECs" select REGMAP +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..2c42f88 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,6 +54,7 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.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 @@ -197,6 +198,7 @@ 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_PCM512x) += snd-soc-pcm512x.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..715aee1 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1369 @@ +/* + * 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 "../../pci/hda/hda_codec.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) { + 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: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + 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_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s fs=%d\n", __func__, fs); + if (50 == fs) + 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; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + int status = 0; + 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_dfs = rt286_set_dai_dfs, +}; + +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[] = { + { "INT33CA", 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

4 3

[alsa-devel] [PATCH 0/4] ASoC: simple-card: DT fix and multi DAI links extension
by Jean-Francois Moine 25 Feb '14

25 Feb '14

This patch series fixes a small problem about node device locks and also extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. Jean-Francois Moine (4): ASoC: simple-card: Fix device node locks ASoC: simple-card: dynamically allocate the DAI link array ASoC: simple-card: accept 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 | 148 +++++++++++++++------ 2 files changed, 137 insertions(+), 45 deletions(-) -- 1.9.0

3 9

[alsa-devel] [PATCH 1/2] ASoC: S3C: Remove invalid file reference
by Sachin Kamat 25 Feb '14

25 Feb '14

Remove file references rendered invalid due to relocation. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- include/linux/platform_data/asoc-s3c.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 9efc04dd255a..709c6f7e2f8c 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/audio.h - * +/* * Copyright (c) 2009 Samsung Electronics Co. Ltd * Author: Jaswinder Singh <jassi.brar(a)samsung.com> * -- 1.7.9.5

2 2

Re: [alsa-devel] ASoC: Intel: Add Haswell/Broadwell IPC
by Dan Carpenter 25 Feb '14

25 Feb '14

Hello Liam Girdwood, The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from Feb 20, 2014, leads to the following imaginary static checker warning: sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() warn: this is always true. sound/soc/intel/sst-haswell-ipc.c 895 /* Handle Immediate reply from DSP Core */ 896 handled = hsw_process_reply(hsw, ipcx); ^^^^^^^^^^^^^^^^^ Returns 1 on success/error and -EIO on error. 897 898 if (handled) { 899 /* clear DONE bit - tell DSP we have completed */ 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, 901 SST_IPCX_DONE, 0); 902 903 /* unmask Done interrupt */ 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 905 SST_IMRX_DONE, 0); 906 } regards, dan carpenter

1 0

[alsa-devel] [patch 2/2] ASoC: sst-haswell-ipc: incorrect sizeof()
by Dan Carpenter 25 Feb '14

25 Feb '14

This should be sizeof(pos) instead of sizeof(&pos). Most likely they are both 8 bytes though so it doesn't often make a difference in real life. Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 552aebf2c8f5..1f1576a9586a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -617,7 +617,7 @@ static void hsw_notification_work(struct work_struct *work) case IPC_POSITION_CHANGED: trace_ipc_notification("DSP stream position changed for", stream->reply.stream_hw_id); - sst_dsp_inbox_read(hsw->dsp, pos, sizeof(&pos)); + sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos)); if (stream->notify_position) stream->notify_position(stream, stream->pdata);

1 0

[alsa-devel] External sound card issue
by Ionica Bizau 25 Feb '14

25 Feb '14

Hi, I have just bought an external sound card: Asus Xonar U7 I typed in terminal: alsamixer Then I wanted to change the sound card (I have connected an external sound card via USB). I pressed F6, selected the external sound card and pressed enter. alsamixer crashed and outputed: cannot load mixer controls: Broken pipe My ALSA information is located at http://www.alsa-project.org/db/?f=1871c086b58a2965d18cfb38483794788187edfb Please see this bug reported in alsa-driver package: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1267866 Tell me what other information you need. Can you fix this bug in 7 days? If you cannot I will return the sound card to the shop.... :-( Thank you! Good luck! I wait for your answer, Ionica Bizau

6 15

[alsa-devel] [PATCH 0/7] ASoC: Intel: Add support for Baytrail SST DSP
by Jarkko Nikula 25 Feb '14

25 Feb '14

This set adds support for Baytrail SST DSP with RT5640 based machine driver. Design is quite similar and is actually inherited from Intel Haswell SST DSP drivers that were posted recently. Reason for another implementation is that there are more differences than similarities between Haswell and Baytrail. First difference comes from the SHIM registers that are 32-bit on Haswell and 64-bit on Baytrail. Also memory map and message mailbox is different. Second difference is message passing and message headers between host and SST DSP. For instance the DSP notifies the host using different SHIM registers and bits than on Haswell. Biggest difference comes from the DSP firmware. It has different IPC messages, stream setup and how DMA pointer is tracked. Currently firmware has less features than Haswell DSP firmware and it doesn't support compress offload and volume controls support. The RT5640 based machine driver is for our customer reference boards but at quick test it appears to work on Asus T100 too. Note currently the SST is the I2S master on this machine driver and codec is set to use I2C BCLK as it's input clock. Jarkko Nikula (7): ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits ASoC: Intel: Add Intel Baytrail SST DSP support ASoC: Intel: Add Intel Baytrail SST DSP IPC support ASoC: Intel: Add Intel Baytrail SST PCM platform driver ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing ASoC: Intel: Add build support for Baytrail SST sound/soc/intel/Kconfig | 14 +- sound/soc/intel/Makefile | 5 + sound/soc/intel/byt-rt5640.c | 194 +++++++++ sound/soc/intel/sst-acpi.c | 17 + sound/soc/intel/sst-baytrail-dsp.c | 372 ++++++++++++++++ sound/soc/intel/sst-baytrail-ipc.c | 866 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 +++ sound/soc/intel/sst-baytrail-pcm.c | 422 ++++++++++++++++++ sound/soc/intel/sst-dsp-priv.h | 1 + sound/soc/intel/sst-dsp.h | 11 + 10 files changed, 1970 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/byt-rt5640.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 -- 1.8.5.3

3 10

[alsa-devel] [PATCH] ASoC: wm8958-dsp: Fix firmware block loading
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

The codec->control_data contains a pointer to the device's regmap struct. But wm8994_bulk_write() expects a pointer to the parent wm8998 device. The issue was introduced in commit d9a7666f ("ASoC: Remove ASoC-specific WM8994 I/O code"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 1974377..7ac2e51 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); -- 1.8.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8996: Replace codec->control_data with wm8996->regmap
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm8996 driver for this by using wm8996->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0330165..e2d0dd7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1608,8 +1608,8 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(wm8996->regmap, false); + regcache_sync(wm8996->regmap); } /* Bypass the MICBIASes for lowest power */ @@ -1620,10 +1620,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -- 1.8.0

2 3

[alsa-devel] [PATCH] ASoC: wm8993: Remove unused pointer in wm8993_remove()
by Christian Engelmayer 25 Feb '14

25 Feb '14

Commit 88b5bdfd (ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data) eliminated the last user of driver data pointer 'wm8993' in function wm8993_remove() - Thus remove it. Detected by Coverity: CID 1186208. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Applies against branch for-next in tree git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/wm8993.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 2ee23a3..f7978b3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1559,8 +1559,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) static int wm8993_remove(struct snd_soc_codec *codec) { - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: da732x: Use da732x->regmap instead of codec->control_data
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the da732x driver for this by using da732x->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/da732x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b65..54db45d 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1487,8 +1487,8 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, da732x_hp_dc_offset_cancellation(codec); - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(da732x->regmap, false); + regcache_sync(da732x->regmap); } else { snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_BOOST_MASK, @@ -1499,7 +1499,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(da732x->regmap, true); da732x_set_charge_pump(codec, DA732X_DISABLE_CP); snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, DA732X_BIAS_DIS); -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: cs42888: Add codec driver support
by Nicolin Chen 25 Feb '14

25 Feb '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ 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> --- .../devicetree/bindings/sound/cs42888.txt | 27 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42888.c | 552 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 209 ++++++++ 5 files changed, 795 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..7736527 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,27 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : phandle to the clock source for MCLK + + - 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 f2383eb..3211488 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_CS42888 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -300,6 +301,10 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate "Cirrus Logic CS42888 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..75fe757 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,7 @@ 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-cs42888-objs := cs42888.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -173,6 +174,7 @@ 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-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o obj-$(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/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8561a25 --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,552 @@ +/* + * Cirrus Logic 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/i2c.h> +#include <linux/module.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 "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_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 cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + 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 *cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum cs42888_enum[] = { + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single), + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single), + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc), + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc), +}; + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", cs42888_enum[0]), + SOC_ENUM("ADC2 Single Ended Mode Switch", cs42888_enum[1]), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", cs42888_enum[2]), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", cs42888_enum[3]), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_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", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_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", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_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" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = 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 = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static struct reg_default cs42888_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 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +static const struct regmap_config cs42888_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42888_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Disable auto-mute */ + regmap_update_bits(cs42888->regmap, CS42888_TXCTL, + CS42888_TXCTL_AMUTE | CS42888_TXCTL_DAC_SZC_MASK, + CS42888_TXCTL_DAC_SZC_SR); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(&i2c->dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, cs42888); + + cs42888->clk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(cs42888->clk)) + dev_warn(&i2c->dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + else + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(&i2c->dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(&i2c->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = devm_regmap_init_i2c(i2c, &cs42888_regmap); + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(&i2c->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(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(&i2c->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 & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(&i2c->dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(&i2c->dev, "found device at %X, revision %X\n", + i2c->addr, val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + ret = snd_soc_register_codec(&i2c->dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(&i2c->dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + + return ret; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c_client) +{ + snd_soc_unregister_codec(&i2c_client->dev); + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(cs42888->clk)) + clk_prepare_enable(cs42888->clk); + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* + * In case the device was put to hard reset during sleep, + * we need to wait 500ns here before any I2C communication + */ + mdelay(5); + + regcache_cache_only(cs42888->regmap, false); + + regcache_sync(cs42888->regmap); + + return 0; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + if (!IS_ERR(cs42888->clk)) + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +static const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .of_match_table = cs42888_of_match, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..6d2cdec --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,209 @@ +/* + * cs42888.h - Cirrus Logic 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 _CS42888_H +#define _CS42888_H + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

5 19

[alsa-devel] [PATCH] ASOC: dapm: add param_fixup callback
by Nikesh＠alsa-project.org 25 Feb '14

25 Feb '14

From: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional callback to modify these params. Change-Id: Iad6ee3951bc4e8b8bc519c62642a2b4bcd949c18 Signed-off-by: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> --- include/sound/soc-dapm.h | 7 ++++--- include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 21 +++++++++++++++++---- 4 files changed, 26 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 68d92e3..e3497ed 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -414,9 +414,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, + struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); @@ -561,7 +562,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ - const struct snd_soc_pcm_stream *params; /* params for dai links */ + struct snd_soc_pcm_stream *params; /* params for dai links */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..401c911 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -875,7 +875,9 @@ struct snd_soc_dai_link { const struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ - const struct snd_soc_pcm_stream *params; + struct snd_soc_pcm_stream *params; + /* optional params re-writing for dai links */ + int (*params_fixup)(struct snd_soc_dapm_widget *w, int event); unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..4f03e88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1469,7 +1469,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1481,7 +1481,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..dc1ef8b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3378,11 +3378,12 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_dai_link *dai_link = w->priv; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; - int ret; + int ret = 0; if (WARN_ON(!config) || WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) @@ -3402,6 +3403,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + if (dai_link && dai_link->params_fixup) { + ret = dai_link->params_fixup(w, event); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: params_fixup for dai link widget failed %d\n", + ret); + goto out; + } + } + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3483,9 +3494,10 @@ out: } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, + struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link) { struct snd_soc_dapm_route routes[2]; struct snd_soc_dapm_widget template; @@ -3517,6 +3529,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; + w->priv = (void *)dai_link; memset(&routes, 0, sizeof(routes)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v7] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 24 Feb '14

24 Feb '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. 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 de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ 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..5085c58 --- /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_create(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

1 0

[alsa-devel] [PATCH v6] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 24 Feb '14

24 Feb '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. 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 | 488 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 505 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 de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ 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..82d5e9d --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,488 @@ +/* + * 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 DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#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 len) +{ + unsigned int 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, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, 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, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command 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(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__, 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, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + 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(dev, "unable to submit urb. OOM!?\n"); +} + +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_create(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); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + 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 = DEVICE_SHORTNAME, + .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

2 3

[alsa-devel] [PATCH 1/6] ASoC: Intel: Add support for Haswell/Broadwell DSP
by Liam Girdwood 24 Feb '14

24 Feb '14

Add support for low level differentiation functions for Haswell and Broadwell SST DSPs. This includes suppoprt for DSP boot and reset, DSP firmware module parsing and DSP memory block map initialisation. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp.c | 60 +++++ sound/soc/intel/sst-dsp.h | 10 + sound/soc/intel/sst-haswell-dsp.c | 517 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 587 insertions(+) create mode 100644 sound/soc/intel/sst-haswell-dsp.c diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 6e22c12..0c129fd 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -27,6 +27,66 @@ #define CREATE_TRACE_POINTS #include <trace/events/intel-sst.h> +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value) +{ + writel(value, addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_write); + +u32 sst_shim32_read(void __iomem *addr, u32 offset) +{ + return readl(addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_read); + +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) +{ + memcpy_toio(addr + offset, &value, sizeof(value)); +} +EXPORT_SYMBOL_GPL(sst_shim32_write64); + +u64 sst_shim32_read64(void __iomem *addr, u32 offset) +{ + u64 val; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + return val; +} +EXPORT_SYMBOL_GPL(sst_shim32_read64); + +static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest, + u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + writel(src[i], dest + i); +} + +static inline void _sst_memcpy_fromio_32(u32 *dest, + const volatile __iomem u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + dest[i] = readl(src + i); +} + +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes) +{ + _sst_memcpy_toio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_toio_32); + +void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest, + void __iomem *src, size_t bytes) +{ + _sst_memcpy_fromio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32); + /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) { diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 3730fd3..608418c 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -189,6 +189,16 @@ u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, u64 mask, u64 value); +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value); +u32 sst_shim32_read(void __iomem *addr, u32 offset); +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value); +u64 sst_shim32_read64(void __iomem *addr, u32 offset); +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes); +void sst_memcpy_fromio_32(struct sst_dsp *sst, + void *dest, void __iomem *src, size_t bytes); + /* DSP reset & boot */ void sst_dsp_reset(struct sst_dsp *sst); int sst_dsp_boot(struct sst_dsp *sst); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c new file mode 100644 index 0000000..12f7317 --- /dev/null +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -0,0 +1,517 @@ +/* + * Intel Haswell SST DSP driver + * + * Copyright (C) 2013, Intel 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. + * + * 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/delay.h> +#include <linux/fs.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/sched.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> + +#include <linux/acpi.h> +#include <acpi/acpi_bus.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-haswell-ipc.h" + +#include <trace/events/hswadsp.h> + +#define SST_HSW_FW_SIGNATURE_SIZE 4 +#define SST_HSW_FW_SIGN "$SST" +#define SST_HSW_FW_LIB_SIGN "$LIB" + +#define SST_WPT_SHIM_OFFSET 0xFB000 +#define SST_LP_SHIM_OFFSET 0xE7000 +#define SST_WPT_IRAM_OFFSET 0xA0000 +#define SST_LP_IRAM_OFFSET 0x80000 + +#define SST_SHIM_PM_REG 0x84 + +#define SST_HSW_IRAM 1 +#define SST_HSW_DRAM 2 +#define SST_HSW_REGS 3 + +struct dma_block_info { + __le32 type; /* IRAM/DRAM */ + __le32 size; /* Bytes */ + __le32 ram_offset; /* Offset in I/DRAM */ + __le32 rsvd; /* Reserved field */ +} __attribute__((packed)); + +struct fw_module_info { + __le32 persistent_size; + __le32 scratch_size; +} __attribute__((packed)); + +struct fw_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */ + __le32 file_size; /* size of fw minus this header */ + __le32 modules; /* # of modules */ + __le32 file_format; /* version of header format */ + __le32 reserved[4]; +} __attribute__((packed)); + +struct fw_module_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */ + __le32 mod_size; /* size of module */ + __le32 blocks; /* # of blocks */ + __le16 padding; + __le16 type; /* codec type, pp lib */ + __le32 entry_point; + struct fw_module_info info; +} __attribute__((packed)); + +static void hsw_free(struct sst_dsp *sst); + +static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + void __iomem *ram; + + /* TODO: allowed module types need to be configurable */ + if (module->type != SST_HSW_MODULE_BASE_FW + && module->type != SST_HSW_MODULE_PCM_SYSTEM + && module->type != SST_HSW_MODULE_PCM + && module->type != SST_HSW_MODULE_PCM_REFERENCE + && module->type != SST_HSW_MODULE_PCM_CAPTURE + && module->type != SST_HSW_MODULE_LPAL) + return 0; + + dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", + module->info.persistent_size, module->info.scratch_size); + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.size = module->info.persistent_size; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.size = module->info.scratch_size; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, + "error: block %d size invalid\n", count); + sst_module_free(mod); + return -EINVAL; + } + + switch (block->type) { + case SST_HSW_IRAM: + ram = dsp->addr.lpe; + block_data.offset = + block->ram_offset + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_HSW_DRAM: + ram = dsp->addr.lpe; + block_data.offset = block->ram_offset; + block_data.type = SST_MEM_DRAM; + break; + default: + dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", + block->type, count); + sst_module_free(mod); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + block_data.data_offset = block_data.data - fw->dma_buf; + + dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + "size 0x%x ==> ram %p offset 0x%x\n", + count, block->type, block->size, ram, + block->ram_offset); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int hsw_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_module *scratch; + struct fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_hsw *hsw = sst_fw->private; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->file_size, header->modules, + header->file_format, sizeof(*header)); + + /* parse each module */ + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + + /* module */ + ret = hsw_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "error: invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + /* allocate persistent/scratch mem regions */ + scratch = sst_mem_block_alloc_scratch(dsp); + if (scratch == NULL) + return -ENOMEM; + + sst_hsw_set_scratch_module(hsw, scratch); + + return 0; +} + +static irqreturn_t hsw_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u32 isr; + int ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + /* Interrupt arrived, check src */ + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + if (isr & SST_ISRX_DONE) { + trace_sst_irq_done(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Done interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, SST_IMRX_DONE); + ret = IRQ_WAKE_THREAD; + } + + if (isr & SST_ISRX_BUSY) { + trace_sst_irq_busy(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Busy interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, SST_IMRX_BUSY); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + return ret; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); + + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + + /* disable DMA finish function for SSP0 & SSP1 */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, + SST_CSR2_SDFD_SSP1); + + /* enable DMA engine 0,1 all channels to access host memory */ + sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + + /* disable all clock gating */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set DSP to RUN */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* lynx point ADSP mem regions */ +static const struct sst_adsp_memregion lp_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */ +}; + +/* wild cat point ADSP mem regions */ +static const struct sst_adsp_memregion wpt_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ +}; + +static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + /* ADSP DRAM & IRAM */ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + return 0; +} + +static u32 hsw_block_get_bit(struct sst_mem_block *block) +{ + u32 bit = 0, shift = 0; + + switch (block->type) { + case SST_MEM_DRAM: + shift = 16; + break; + case SST_MEM_IRAM: + shift = 6; + break; + default: + return 0; + } + + bit = 1 << (block->index + shift); + + return bit; +} + +/* enable 32kB memory block - locks held by caller */ +static int hsw_block_enable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (block->users++ > 0) + return 0; + + dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* wait 18 DSP clock ticks */ + udelay(10); + + return 0; +} + +/* disable 32kB memory block - locks held by caller */ +static int hsw_block_disable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (--block->users > 0) + return 0; + + dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static struct sst_block_ops sst_hsw_ops = { + .enable = hsw_block_enable, + .disable = hsw_block_disable, +}; + +static int hsw_enable_shim(struct sst_dsp *sst) +{ + int tries = 10; + u32 reg; + + /* enable shim */ + reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); + writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); + if (reg != 0xffffffff) + return 0; + + msleep(1); + } + + return -ENODEV; +} + +static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_LYNX_POINT: + region = lp_region; + region_count = ARRAY_SIZE(lp_region); + sst->addr.iram_offset = SST_LP_IRAM_OFFSET; + sst->addr.shim_offset = SST_LP_SHIM_OFFSET; + break; + case SST_DEV_ID_WILDCAT_POINT: + region = wpt_region; + region_count = ARRAY_SIZE(wpt_region); + sst->addr.iram_offset = SST_WPT_IRAM_OFFSET; + sst->addr.shim_offset = SST_WPT_SHIM_OFFSET; + break; + default: + dev_err(dev, "error: failed to get mem resources\n"); + return ret; + } + + ret = hsw_acpi_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "error: failed to map resources\n"); + return ret; + } + + /* enable the DSP SHIM */ + ret = hsw_enable_shim(sst); + if (ret < 0) { + dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); + return ret; + } + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, + (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, &sst_hsw_ops, j, sst); + offset += size; + } + } + + /* set default power gating mask */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static void hsw_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); +} + +struct sst_ops haswell_ops = { + .reset = hsw_reset, + .boot = hsw_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = hsw_irq, + .init = hsw_init, + .free = hsw_free, + .parse_fw = hsw_parse_fw_image, +}; -- 1.8.3.2

4 21

[alsa-devel] [PATCH] ALSA: hda - Add a fixup for HP Folio 13 mute LED
by Takashi Iwai 24 Feb '14

24 Feb '14

HP Folio 13 may have a broken BIOS that doesn't set up the mute LED GPIO properly, and the driver guesses it wrongly, too. Add a new fixup entry for setting the GPIO pin statically for this laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2f11bf8155c..3bc29c9b2529 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -98,6 +98,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_HP_LED_GPIO10, STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, @@ -2130,6 +2131,17 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, } } +static void stac92hd83xxx_fixup_hp_led_gpio10(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0x10; /* GPIO4 */ + spec->default_polarity = 0; + } +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2624,6 +2636,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP, }, + [STAC_HP_LED_GPIO10] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_led_gpio10, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, [STAC_92HD83XXX_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_headset_jack, @@ -2702,6 +2720,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1888, "HP Envy Spectre", STAC_HP_ENVY_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1899, + "HP Folio 13", STAC_HP_LED_GPIO10), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, "HP Folio", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18F8, -- 1.9.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Avoid codec D3 for keeping mute LED up on Lenovo Yxx0
by Takashi Iwai 24 Feb '14

24 Feb '14

The GPIO line used for the mute LED control on Lenovo Yxx0 laptops is cleared unexpectedly when the codec goes to D3, typically by power-saving. For avoiding it, add a power filter in the fixup. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3829cb5ebc2b..817a52efe750 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4832,6 +4832,17 @@ static void alc662_led_gpio1_mute_hook(void *private_data, int enabled) spec->gpio_led); } +/* avoid D3 for keeping GPIO up */ +static unsigned int gpio_led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + if (nid == codec->afg && power_state == AC_PWRST_D3 && spec->gpio_led) + return AC_PWRST_D0; + return power_state; +} + static void alc662_fixup_led_gpio1(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: Use priority list for managing device list
by Takashi Iwai 24 Feb '14

24 Feb '14

Basically, the device type specifies the priority of the device to be registered / freed, too. However, the priority value isn't well utilized but only it's checked as a group. This results in inconsistent register and free order (where each of them should be in reversed direction). This patch simplifies the device list management code by simply inserting a list entry at creation time in an incremental order for the priority value. Since we can just follow the link for register, disconnect and free calls, we don't have to specify the group; so the whole enum definitions are also simplified as well. The visible change to outside is that the priorities of some object types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object is registered before others (control, PCM, etc) and, in return, released after others. Similarly, SNDRV_DEV_CODEC is in a lower priority than SNDRV_DEV_BUS for ensuring the dependency. Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 28 ++++++++-------------- sound/core/device.c | 66 +++++++++++++++++++++++++++++----------------------- sound/core/init.c | 14 +++-------- 3 files changed, 50 insertions(+), 58 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index a3e3e89b63b6..5927378dc692 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -46,24 +46,22 @@ struct completion; /* device allocation stuff */ -#define SNDRV_DEV_TYPE_RANGE_SIZE 0x1000 - +/* type of the object used in snd_device_*() + * this also defines the calling order + */ enum snd_device_type { - SNDRV_DEV_TOPLEVEL = 0, - SNDRV_DEV_CONTROL = 1, - SNDRV_DEV_LOWLEVEL_PRE = 2, - SNDRV_DEV_LOWLEVEL_NORMAL = 0x1000, + SNDRV_DEV_LOWLEVEL, + SNDRV_DEV_CONTROL, + SNDRV_DEV_INFO, + SNDRV_DEV_BUS, + SNDRV_DEV_CODEC, SNDRV_DEV_PCM, + SNDRV_DEV_COMPRESS, SNDRV_DEV_RAWMIDI, SNDRV_DEV_TIMER, SNDRV_DEV_SEQUENCER, SNDRV_DEV_HWDEP, - SNDRV_DEV_INFO, - SNDRV_DEV_BUS, - SNDRV_DEV_CODEC, SNDRV_DEV_JACK, - SNDRV_DEV_COMPRESS, - SNDRV_DEV_LOWLEVEL = 0x2000, }; enum snd_device_state { @@ -72,12 +70,6 @@ enum snd_device_state { SNDRV_DEV_DISCONNECTED, }; -enum snd_device_cmd { - SNDRV_DEV_CMD_PRE, - SNDRV_DEV_CMD_NORMAL, - SNDRV_DEV_CMD_POST, -}; - struct snd_device; struct snd_device_ops { @@ -321,7 +313,7 @@ int snd_device_register_all(struct snd_card *card); int snd_device_disconnect(struct snd_card *card, void *device_data); int snd_device_disconnect_all(struct snd_card *card); int snd_device_free(struct snd_card *card, void *device_data); -int snd_device_free_all(struct snd_card *card, enum snd_device_cmd cmd); +int snd_device_free_all(struct snd_card *card); /* isadma.c */ diff --git a/sound/core/device.c b/sound/core/device.c index 856bfdc79cee..53291ff21a09 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -45,6 +45,7 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops) { struct snd_device *dev; + struct list_head *p; if (snd_BUG_ON(!card || !device_data || !ops)) return -ENXIO; @@ -53,17 +54,36 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, dev_err(card->dev, "Cannot allocate device, type=%d\n", type); return -ENOMEM; } + INIT_LIST_HEAD(&dev->list); dev->card = card; dev->type = type; dev->state = SNDRV_DEV_BUILD; dev->device_data = device_data; dev->ops = ops; - list_add(&dev->list, &card->devices); /* add to the head of list */ + + /* insert the entry in an incrementally sorted list */ + list_for_each_prev(p, &card->devices) { + struct snd_device *pdev = list_entry(p, struct snd_device, list); + if ((unsigned int)pdev->type <= (unsigned int)type) + break; + } + + list_add(&dev->list, p); return 0; } - EXPORT_SYMBOL(snd_device_new); +static struct snd_device *look_for_dev(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + list_for_each_entry(dev, &card->devices, list) + if (dev->device_data == device_data) + return dev; + + return NULL; +} + /** * snd_device_free - release the device from the card * @card: the card instance @@ -82,9 +102,8 @@ int snd_device_free(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { /* unlink */ list_del(&dev->list); if (dev->state == SNDRV_DEV_REGISTERED && @@ -103,7 +122,6 @@ int snd_device_free(struct snd_card *card, void *device_data) device_data, __builtin_return_address(0)); return -ENXIO; } - EXPORT_SYMBOL(snd_device_free); /** @@ -125,9 +143,8 @@ int snd_device_disconnect(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { if (dev->state == SNDRV_DEV_REGISTERED && dev->ops->dev_disconnect) { if (dev->ops->dev_disconnect(dev)) @@ -162,9 +179,8 @@ int snd_device_register(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { if (dev->state == SNDRV_DEV_BUILD && dev->ops->dev_register) { if ((err = dev->ops->dev_register(dev)) < 0) return err; @@ -177,7 +193,6 @@ int snd_device_register(struct snd_card *card, void *device_data) snd_BUG(); return -ENXIO; } - EXPORT_SYMBOL(snd_device_register); /* @@ -212,7 +227,7 @@ int snd_device_disconnect_all(struct snd_card *card) if (snd_BUG_ON(!card)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { + list_for_each_entry_reverse(dev, &card->devices, list) { if (snd_device_disconnect(card, dev->device_data) < 0) err = -ENXIO; } @@ -223,24 +238,17 @@ int snd_device_disconnect_all(struct snd_card *card) * release all the devices on the card. * called from init.c */ -int snd_device_free_all(struct snd_card *card, enum snd_device_cmd cmd) +int snd_device_free_all(struct snd_card *card) { - struct snd_device *dev; - int err; - unsigned int range_low, range_high, type; + struct snd_device *dev, *next; + int ret = 0; if (snd_BUG_ON(!card)) return -ENXIO; - range_low = (unsigned int)cmd * SNDRV_DEV_TYPE_RANGE_SIZE; - range_high = range_low + SNDRV_DEV_TYPE_RANGE_SIZE - 1; - __again: - list_for_each_entry(dev, &card->devices, list) { - type = (unsigned int)dev->type; - if (type >= range_low && type <= range_high) { - if ((err = snd_device_free(card, dev->device_data)) < 0) - return err; - goto __again; - } + list_for_each_entry_safe_reverse(dev, next, &card->devices, list) { + int err = snd_device_free(card, dev->device_data); + if (err < 0) + ret = err; } - return 0; + return ret; } diff --git a/sound/core/init.c b/sound/core/init.c index b7085eb19079..9e7f17b72fb6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -266,7 +266,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid, return 0; __error_ctl: - snd_device_free_all(card, SNDRV_DEV_CMD_PRE); + snd_device_free_all(card); __error: put_device(&card->card_dev); return err; @@ -454,16 +454,8 @@ static int snd_card_do_free(struct snd_card *card) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif - if (snd_device_free_all(card, SNDRV_DEV_CMD_PRE) < 0) { - dev_err(card->dev, "unable to free all devices (pre)\n"); - /* Fatal, but this situation should never occur */ - } - if (snd_device_free_all(card, SNDRV_DEV_CMD_NORMAL) < 0) { - dev_err(card->dev, "unable to free all devices (normal)\n"); - /* Fatal, but this situation should never occur */ - } - if (snd_device_free_all(card, SNDRV_DEV_CMD_POST) < 0) { - dev_err(card->dev, "unable to free all devices (post)\n"); + if (snd_device_free_all(card) < 0) { + dev_err(card->dev, "unable to free all devices\n"); /* Fatal, but this situation should never occur */ } if (card->private_free) -- 1.8.5.2

1 1

[alsa-devel] [PATCH] ASoC: add data field into snd_soc_jack_gpio
by xiangxiao 24 Feb '14

24 Feb '14

so callback could get the context data as needed Change-Id: I5542613ccf9881deb672e71d0f0c4f603ad761fd Signed-off-by: xiangxiao <xiaoxiang(a)xiaomi.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-jack.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 21038e0..a789a29 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -613,7 +613,8 @@ struct snd_soc_jack_gpio { struct snd_soc_jack *jack; struct delayed_work work; - int (*jack_status_check)(void); + void *data; + int (*jack_status_check)(void *data); }; struct snd_soc_jack { diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b12fce6..b903f82 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) report = 0; if (gpio->jack_status_check) - report = gpio->jack_status_check(); + report = gpio->jack_status_check(gpio->data); snd_soc_jack_report(jack, report, gpio->report); } -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: Remove unneeded 'out' label
by Fabio Estevam 24 Feb '14

24 Feb '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Instead of jumping to the 'out' label, just return the error code immediately. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-pcm-fiq.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 6553202..7abf6a0 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -270,18 +270,17 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) - goto out; + return ret; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) - goto out; + return ret; } -out: - return ret; + return 0; } static int ssi_irq = 0; -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: core: add snd_soc_dai_set_tdm_slot_xlate()
by Xiubo Li 24 Feb '14

24 Feb '14

The parameters of rx_mask and tx_mask has no use for snd_soc_dai_set_tdm_slot(), because they could be generated by {XXX_ .}of_xlate_tdm_slot_mask(). This patch add snd_soc_dai_set_tdm_slot_xlate() which will replace the snd_soc_dai_set_tdm_slot() eventually. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 33 ++++++++++++++++++++++++++++----- 2 files changed, 31 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d86e0fc..68569ee 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -110,6 +110,9 @@ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); +int snd_soc_dai_set_tdm_slot_xlate(struct snd_soc_dai *dai, + unsigned int slots, + unsigned int slot_width); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0911856..e5a535b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3687,19 +3687,20 @@ static int snd_soc_of_xlate_tdm_slot_mask(unsigned int slots, } /** - * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * snd_soc_dai_set_tdm_slot_xlate - configure DAI TDM with of xlate. * @dai: DAI - * @tx_mask: bitmask representing active TX slots. - * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ -int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +int snd_soc_dai_set_tdm_slot_xlate(struct snd_soc_dai *dai, + unsigned int slots, + unsigned int slot_width) { + unsigned int tx_mask, rx_mask; + if (dai->driver && dai->driver->ops->of_xlate_tdm_slot_mask) dai->driver->ops->of_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); @@ -3712,6 +3713,28 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, else return -ENOTSUPP; } +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot_xlate); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + if (dai->driver && dai->driver->ops->set_tdm_slot) + return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); + else + return -ENOTSUPP; +} EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: da732x: Remove leftover cache size setting
by Mark Brown 24 Feb '14

24 Feb '14

From: Mark Brown <broonie(a)linaro.org> The da732x driver no longer uses the ASoC level register cache but the cache size setting had been left in the driver by mistake. Remove it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/da732x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b65..8e2c74d 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1554,7 +1554,6 @@ static struct snd_soc_codec_driver soc_codec_dev_da732x = { .dapm_routes = da732x_dapm_routes, .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), .set_pll = da732x_set_dai_pll, - .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), }; static int da732x_i2c_probe(struct i2c_client *i2c, -- 1.9.0.rc3

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Add PCM512x driver
by Mark Brown 24 Feb '14

24 Feb '14

From: Mark Brown <broonie(a)linaro.org> The PCM512x devices are a family of monolithic CMOS integrated circuits that include a stereo digital-to-analog converter and additional support circuitry. This is an initial driver which supports some core functionality for the device which covers common use cases but does not cover all features. Currently only slave clocking modes with automatic clock configuration are supported and most of the DSP configuration for the device is not enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../devicetree/bindings/sound/pcm512x.txt | 30 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm512x.c | 671 +++++++++++++++++++++ sound/soc/codecs/pcm512x.h | 142 +++++ 5 files changed, 849 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt new file mode 100644 index 000000000000..faff75e64573 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -0,0 +1,30 @@ +PCM512x audio CODECs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : One of "ti,pcm5121" or "ti,pcm5122" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the + device, as covered in bindings/regulator/regulator.txt + +Optional properties: + + - clocks : A clock specifier for the clock connected as SCLK. If this + is absent the device will be configured to clock from BCLK. + +Example: + + pcm5122: pcm5122@4c { + compatible = "ti,pcm5122"; + reg = <0x4c>; + + AVDD-supply = <&reg_3v3_analog>; + DVDD-supply = <&reg_1v8>; + CPVDD-supply = <&reg_3v3>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0bdcc5abe83..28faa2d0eaef 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -324,6 +325,9 @@ config SND_SOC_PCM1792A config SND_SOC_PCM3008 tristate +config SND_SOC_PCM512x + tristate "Texas Instruments PCM512x CODECs" + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..d3b536fc075d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm512x-objs := pcm512x.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -179,6 +180,7 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(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-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(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/pcm512x.c b/sound/soc/codecs/pcm512x.c new file mode 100644 index 000000000000..886e530ad2d2 --- /dev/null +++ b/sound/soc/codecs/pcm512x.c @@ -0,0 +1,671 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "pcm512x.h" + +#define PCM512x_NUM_SUPPLIES 3 +static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "CPVDD", +}; + +struct pcm512x_priv { + struct regmap *regmap; + struct clk *sclk; + struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES]; + struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES]; +}; + +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define PCM512x_REGULATOR_EVENT(n) \ +static int pcm512x_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \ + supply_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + regcache_mark_dirty(pcm512x->regmap); \ + regcache_cache_only(pcm512x->regmap, true); \ + } \ + return 0; \ +} + +PCM512x_REGULATOR_EVENT(0) +PCM512x_REGULATOR_EVENT(1) +PCM512x_REGULATOR_EVENT(2) + +static const struct reg_default pcm512x_reg_defaults[] = { + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, +}; + +static bool pcm512x_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_RESET: + case PCM512x_POWER: + case PCM512x_MUTE: + case PCM512x_PLL_EN: + case PCM512x_SPI_MISO_FUNCTION: + case PCM512x_DSP: + case PCM512x_GPIO_EN: + case PCM512x_BCLK_LRCLK_CFG: + case PCM512x_DSP_GPIO_INPUT: + case PCM512x_MASTER_MODE: + case PCM512x_PLL_REF: + case PCM512x_PLL_COEFF_0: + case PCM512x_PLL_COEFF_1: + case PCM512x_PLL_COEFF_2: + case PCM512x_PLL_COEFF_3: + case PCM512x_PLL_COEFF_4: + case PCM512x_DSP_CLKDIV: + case PCM512x_DAC_CLKDIV: + case PCM512x_NCP_CLKDIV: + case PCM512x_OSR_CLKDIV: + case PCM512x_MASTER_CLKDIV_1: + case PCM512x_MASTER_CLKDIV_2: + case PCM512x_FS_SPEED_MODE: + case PCM512x_IDAC_1: + case PCM512x_IDAC_2: + case PCM512x_ERROR_DETECT: + case PCM512x_I2S_1: + case PCM512x_I2S_2: + case PCM512x_DAC_ROUTING: + case PCM512x_DSP_PROGRAM: + case PCM512x_CLKDET: + case PCM512x_AUTO_MUTE: + case PCM512x_DIGITAL_VOLUME_1: + case PCM512x_DIGITAL_VOLUME_2: + case PCM512x_DIGITAL_VOLUME_3: + case PCM512x_DIGITAL_MUTE_1: + case PCM512x_DIGITAL_MUTE_2: + case PCM512x_DIGITAL_MUTE_3: + case PCM512x_GPIO_OUTPUT_1: + case PCM512x_GPIO_OUTPUT_2: + case PCM512x_GPIO_OUTPUT_3: + case PCM512x_GPIO_OUTPUT_4: + case PCM512x_GPIO_OUTPUT_5: + case PCM512x_GPIO_OUTPUT_6: + case PCM512x_GPIO_CONTROL_1: + case PCM512x_GPIO_CONTROL_2: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static bool pcm512x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_PLL_EN: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); + +static const char *pcm512x_dsp_program_texts[] = { + "FIR interpolation with de-emphasis", + "Low latency IIR with de-emphasis", + "High attenuation with de-emphasis", + "Ringing-less low latency FIR", +}; + +static const unsigned int pcm512x_dsp_program_values[] = { + 1, + 2, + 3, + 5, + 7, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); + +static const char *pcm512x_clk_missing_text[] = { + "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" +}; + +static const struct soc_enum pcm512x_clk_missing = + SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 7, pcm512x_clk_missing_text); + +static const char *pcm512x_autom_text[] = { + "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" +}; + +static const struct soc_enum pcm512x_autom_l = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 7, + pcm512x_autom_text); + +static const struct soc_enum pcm512x_autom_r = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 7, + pcm512x_autom_text); + +static const char *pcm512x_ramp_rate_text[] = { + "1 sample/update", "2 samples/update", "4 samples/update", + "Immediate" +}; + +static const struct soc_enum pcm512x_vndf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vnuf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vedf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const char *pcm512x_ramp_step_text[] = { + "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" +}; + +static const struct soc_enum pcm512x_vnds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_vnus = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_veds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct snd_kcontrol_new pcm512x_controls[] = { +SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, + PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, + PCM512x_RQMR_SHIFT, 1, 1), + +SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), +SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program), + +SOC_ENUM("Clock Missing Period", pcm512x_clk_missing), +SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l), +SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r), +SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3, + PCM512x_ACTL_SHIFT, 1, 0), +SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT, + PCM512x_AMLR_SHIFT, 1, 0), + +SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf), +SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds), +SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), +SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), +SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), +SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), +}; + +static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { + { "DACL", NULL, "Playback" }, + { "DACR", NULL, "Playback" }, + + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +static int pcm512x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to remove standby: %d\n", + ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(codec->dev, "Failed to request standby: %d\n", + ret); + return ret; + } + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_dai_driver pcm512x_dai = { + .name = "pcm512x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver pcm512x_codec_driver = { + .set_bias_level = pcm512x_set_bias_level, + .idle_bias_off = true, + + .controls = pcm512x_controls, + .num_controls = ARRAY_SIZE(pcm512x_controls), + .dapm_widgets = pcm512x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets), + .dapm_routes = pcm512x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), +}; + +static const struct regmap_config pcm512x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = pcm512x_readable, + .volatile_reg = pcm512x_volatile, + + .max_register = PCM512x_MAX_REGISTER, + .reg_defaults = pcm512x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static int pcm512x_probe(struct device *dev, struct regmap *regmap) +{ + struct pcm512x_priv *pcm512x; + int i, ret; + + pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL); + if (!pcm512x) + return -ENOMEM; + + dev_set_drvdata(dev, pcm512x); + pcm512x->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) + pcm512x->supplies[i].supply = pcm512x_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0; + pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1; + pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { + ret = regulator_register_notifier(pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the device, verifying I/O in the process for I2C */ + ret = regmap_write(regmap, PCM512x_RESET, + PCM512x_RSTM | PCM512x_RSTR); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + ret = regmap_write(regmap, PCM512x_RESET, 0); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + pcm512x->sclk = devm_clk_get(dev, NULL); + if (IS_ERR(pcm512x->sclk)) { + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + dev_info(dev, "No SCLK, using BCLK: %ld\n", + PTR_ERR(pcm512x->sclk)); + + /* Disable reporting of missing SCLK as an error */ + regmap_update_bits(regmap, PCM512x_ERROR_DETECT, + PCM512x_IDCH, PCM512x_IDCH); + + /* Switch PLL input to BCLK */ + regmap_update_bits(regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF); + } else { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + /* Default to standby mode */ + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(dev, "Failed to request standby: %d\n", + ret); + goto err_clk; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &pcm512x_codec_driver, + &pcm512x_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_clk: + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); +err: + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + return ret; +} + +static void pcm512x_remove(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); +} + +static int pcm512x_suspend(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, PCM512x_RQPD); + if (ret != 0) { + dev_err(dev, "Failed to request power down: %d\n", ret); + return ret; + } + + ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to disable supplies: %d\n", ret); + return ret; + } + + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + + return 0; +} + +static int pcm512x_resume(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(pcm512x->sclk)) { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(pcm512x->regmap, false); + ret = regcache_sync(pcm512x->regmap); + if (ret != 0) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, 0); + if (ret != 0) { + dev_err(dev, "Failed to remove power down: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct dev_pm_ops pcm512x_pm_ops = { + SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) +}; + +#if IS_ENABLED(CONFIG_I2C) +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +static int __init pcm512x_modinit(void) +{ + int ret = 0; + +#if IS_ENABLED(CONFIG_I2C) + ret = i2c_add_driver(&pcm512x_i2c_driver); + if (ret) { + printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&pcm512x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(pcm512x_modinit); + +static void __exit pcm512x_exit(void) +{ +#if IS_ENABLED(CONFIG_I2C) + i2c_del_driver(&pcm512x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&pcm512x_spi_driver); +#endif +} +module_exit(pcm512x_exit); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h new file mode 100644 index 000000000000..b2f518ecb35c --- /dev/null +++ b/sound/soc/codecs/pcm512x.h @@ -0,0 +1,142 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * 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. + * + * 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. + */ + +#ifndef _SND_SOC_PCM512X +#define _SND_SOC_PCM512X + +#define PCM512x_PAGE_0_BASE 0 + +#define PCM512x_PAGE 0 + +#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) +#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) +#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) +#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) +#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) + +/* Page 0, Register 1 - reset */ +#define PCM512x_RSTR (1 << 0) +#define PCM512x_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define PCM512x_RQPD (1 << 0) +#define PCM512x_RQPD_SHIFT 0 +#define PCM512x_RQST (1 << 4) +#define PCM512x_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define PCM512x_PLCE (1 << 0) +#define PCM512x_RLCE_SHIFT 0 +#define PCM512x_PLCK (1 << 4) +#define PCM512x_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define PCM512x_SDSL (1 << 0) +#define PCM512x_SDSL_SHIFT 0 +#define PCM512x_DEMP (1 << 4) +#define PCM512x_DEMP_SHIFT 4 + +/* Page 0, Register 13 - PLL reference */ +#define PCM512x_SREF (1 << 4) + +/* Page 0, Register 37 - Error detection */ +#define PCM512x_IPLK (1 << 0) +#define PCM512x_DCAS (1 << 1) +#define PCM512x_IDCM (1 << 2) +#define PCM512x_IDCH (1 << 3) +#define PCM512x_IDSK (1 << 4) +#define PCM512x_IDBK (1 << 5) +#define PCM512x_IDFS (1 << 6) + +/* Page 0, Register 42 - DAC routing */ +#define PCM512x_AUPR_SHIFT 0 +#define PCM512x_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define PCM512x_ATMR_SHIFT 0 +#define PCM512x_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define PCM512x_VNDF_SHIFT 6 +#define PCM512x_VNDS_SHIFT 4 +#define PCM512x_VNUF_SHIFT 2 +#define PCM512x_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define PCM512x_VEDF_SHIFT 6 +#define PCM512x_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define PCM512x_ACTL_SHIFT 2 +#define PCM512x_AMLE_SHIFT 1 +#define PCM512x_AMLR_SHIFT 0 + +#endif -- 1.9.rc1

3 12

[alsa-devel] [PATCH] ASoC: delay the initial jack detect by debounce_time
by xiangxiao 24 Feb '14

24 Feb '14

so the hardware could get time to initialize and debounce Change-Id: Iebeffa9bac6c4012ee8f9645771c13b28fe5ae80 Signed-off-by: xiangxiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 23d43da..b12fce6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_export(gpios[i].gpio, false); /* Update initial jack status */ - snd_soc_jack_gpio_detect(&gpios[i]); + schedule_delayed_work(&gpios[i].work, + msecs_to_jiffies(gpios[i].debounce_time)); } return 0; -- 1.8.1.2

2 1

Re: [alsa-devel] [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 23 Feb '14

23 Feb '14

Dear Takashi, Firstly: thank you so much for taking the time to do this! I am very, very grateful! You asked for the following: $ cat /proc/asound/card2/#clock Sampling rate: 48000 Clock Source: Internal (MSU-dest: 5) $ cat /proc/asound/card2/#firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x000D6C0A009D269F Model ID: 0x01 Model Rev: 1 Firmware Date: 20050905 Firmware Time: 091022 Firmware ID: 0xA Firmware Ver: 20 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20040915 Loader Time: 175644 $ cat /proc/asound/card2/#formation Output Stream from device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 0 0 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 0 0 Input Stream to device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 0 0 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 0 0 $ cat /proc/asound/card2/#meter Analog In 1: 1328896 Analog In 2: 1216256 Analog In 3: 1326097920 Analog In 4: 8012800 Stream In 1: 0 Stream In 2: 0 Stream In 3: 0 Stream In 4: 0 Analog Out 1: 0 Analog Out 2: 0 Analog Out 3: 0 Analog Out 4: 0 Now, my experience: I am testing on a new, well provisioned PC, freshly loaded with Ubuntu 13.10. My firewire card has a Texas Instruments chipset, and my device is the Ozonic. I install via DKMS, as per your instructions, and then sudo modprobe snd-bebob. The snd_firewire_lib and snd_bebob modules are now listed in the output of lsmod. The device has installed as card2 (hw:2,0). I can now test MIDI input with: $ amidi -p hw:2,0,0 -d ... which satisfies me that all MIDI output from the various controllers on the Ozonic is being received. I have not yet tried to send any MIDI data to the device. I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as ALSA input and output options. However, no audio input is shown when I try to monitor the audio input, and trying to play generated audio (e.g. DTMF tones) results in the message: "Error while opening sound device. Please check the output device settings and the project sample rate." I have repeated the experiment at all supported sampling rates (44100, 48000, 88200 & 96000 Hz), and obtained the same result in each case. I then close Audacity and try: $aplay -D hw:2,0 test.wav (where test.wav has been saved from Audacity) ... and this responds with: aplay: main:722: audio open error: Input/output error Based on my experiences with the Ozonic under FFADO, I might easily have resigned myself to thinking that full functionality will only be apparent once a mixer is implemented. I had hacked together the "unmute-ozonic" utility under FFADO in order to overcome the need the need to dual boot in order to get I/O. However, the fact that an error message is returned upon trying to open the Ozonic's output makes me think that this is another problem entirely. UPDATE: I have just this moment been able to do the following: $ aplay -D hw:2,0 Desktop/test.wav Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: set_params:1239: Channels count not available ... which appears in order to me, except that no audio can be heard. I assume this is due to the lack of a mixer. I will retrace my steps to try and figure out why it is now apparently working. However, I now open Audacity again, and the first thing I notice is that my pass-through audio immediately disappears (I have a sound source playing into the unbalanced inputs throughout). Audacity responds exactly the same as before. I close Audacity and aplay can no longer open hw:2,0 (I/O error as before). Now I modprobe -r the modules (and lsmod to confirm they're gone), cycle the Ozonic's power off and on, and the audio pass-through has reappeared! I also see that lsmod lists both snd_firewire_lib & snd_bebob as having reloaded themselves automatically. My testing is probably not particularly rigorous, and I would sincerely appreciate your advice on what steps to follow in order to verify your driver's functionality. In particular, you mention "(to control mixer channels please use FFADO upstream)". I have not yet set up FFADO on this machine, but I also see under "Note": "- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' and 'jackd with Firewire (FFADO) backend'. Both of them try connecting to the device when another is running." Would you please explain? How should one rely on the "upstream" mixer functionality while keeping jackd out of ALSA's way? Once again, thank you sincerely for doing this. I, for one, feel like a kid on Christmas Eve! Kind regards, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Monday, October 14, 2013 5:17 AM To: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net; linux1394-devel(a)lists.sourceforge.net; linux-audio-dev(a)lists.linuxaudio.org Subject: [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices Hi all, This is a call for testing my ALSA driver for Fireworks/BeBoB based devices. Please test 'snd-fireworks' for Fireworks and 'snd-bebob' for BeBoB if you have some devices listed in the end of this mail. [snip] Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp ------------------------------------------------------------------------------ October Webinars: Code for Performance Free Intel webinars can help you accelerate application performance. Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from the latest Intel processors and coprocessors. See abstracts and register > http://pubads.g.doubleclick.net/gampad/clk?id=60134071&iu=/4140/ostg.clktrk _______________________________________________ FFADO-devel mailing list FFADO-devel(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/ffado-devel ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

2 1

Re: [alsa-devel] 答复: [PATCH] ASoC: pass snd_soc_jack_gpio to jack_status_check callback
by Mark Brown 23 Feb '14

23 Feb '14

On Wed, Feb 19, 2014 at 02:20:07PM +0000, 肖翔 wrote: > The new parameter isn't used to get the internal field of snd_soc_jack_gpio. > the usage is very like work_struct, here is the demo code: > struct xxx_priv { > truct snd_soc_jack_gpio gpio; > }; > > static int xxx_ jack_status_check(struct snd_soc_jack_gpio *gpio) > { > struct xxx_priv *xxx = container_of(gpio, struct xxx_priv, gpio); > ...... > } > Without this parameter, codec driver has to save xxx_priv as a global variable. OK, this makes sense as a reason for passing the argument in however how about instead of passing in the structure adding a void * to it which gets passed as the argument instead? The user can then assign it to point to the data they want directly without it looking like they should be using the private data for the core code.

2 1

[alsa-devel] [PATCH v3 0/3] Add TDM slot info OF parsing supports
by Xiubo Li 23 Feb '14

23 Feb '14

Change in v3: - Add .of_xlate_tdm_slot_mask(). - Add snd_soc_of_xlate_tdm_slot_mask(). - Remove TDM slot masks OF parsing... - Split the TDM infomation into two properties. Xiubo Li (3): ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports .../devicetree/bindings/sound/simple-card.txt | 5 ++ .../devicetree/bindings/sound/tdm-slot.txt | 20 ++++++++ include/sound/simple_card.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc.h | 3 ++ sound/soc/generic/simple-card.c | 18 +++++++ sound/soc/soc-core.c | 59 ++++++++++++++++++++++ 7 files changed, 109 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt -- 1.8.4

2 4

[alsa-devel] [PATCH v6 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 23 Feb '14

23 Feb '14

Hi, This series adds master clock and regulators to the tlv320aic32x4 driver. In v6 I added error handling for the regulator setup. master clock is required for the driver now but still optional in the DT bindings. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Asoc: codecs: Remove tlv320aic32x4 from compatibles of tlv320aic3x .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 177 +++++++++++++++++++-- 3 files changed, 173 insertions(+), 17 deletions(-) -- 1.8.5.3

3 8

[alsa-devel] [PATCH 029/102] ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL()
by Takashi Iwai 23 Feb '14

23 Feb '14

Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/adau1373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 46b1595d7c50..5765c224cd9a 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static const struct soc_enum adau1373_decimator_enum = - SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); +static SOC_ENUM_SINGLE_DECL(adau1373_decimator_enum, + 0, 0, adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); -- 1.8.5.2

2 109

[alsa-devel] [PATCH] ASoC: pxa: Remove superfluous locking
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

The locking here was added in commit 71a295602e ("ASoC: Lock the CODEC in PXA external jack controls") to protect the DAPM changes that are made inside of ${board}_ext_control() against concurrent updates. The ASoC core was updated in commit a73fb2df01 ("ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex") to use a card wide lock rather the CODEC mutex to protect DAPM operations. We now have proper locking inside ${board}_ext_control() itself, so taking the CODEC lock can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/magician.c | 4 ---- sound/soc/pxa/poodle.c | 4 ---- sound/soc/pxa/spitz.c | 4 ---- sound/soc/pxa/tosa.c | 4 ---- 5 files changed, 20 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 9d9c8ad..b4db3b9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -103,13 +103,9 @@ static int corgi_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ corgi_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 31242be..41ab6678 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -77,13 +77,9 @@ static int magician_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ magician_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index c93e138..27c6c03 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -76,13 +76,9 @@ static int poodle_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ poodle_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 04dbb5a..a3a13c9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -113,13 +113,9 @@ static int spitz_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ spitz_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2a4b438..cead165 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -84,13 +84,9 @@ static int tosa_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ tosa_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: sta32x: Fix cache sync
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

codec->control_data contains a pointer to the regmap struct of the device, not to the device private data. Use snd_soc_codec_get_drvdata() instead. The issue was introduced in commit 29fdf4fbbe ("ASoC: sta32x: Convert to regmap"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 42c5b45..ea78c17 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; -- 1.8.0

2 1

[alsa-devel] No sound from Headphone Jack over Lenovo Docking Station with T440p Laptop
by Nicolas Kalkhof 22 Feb '14

22 Feb '14

Hello, I have a Lenovo ThinkPad T440p laptop connected to an Ultra Docking Station. Sound works fine over Laptop Speakers and the 3,5mm Headphone Jack. However that is not the case when I connect the T440p to the Docking station Type 40A10065EU and use the Headphone Jack at the Dock. I've tried loading the alsa module with the "model=thinkpad" or "model=lenovo-dock" without success. My Distro is Gentoo and the Kernel is 3.13-rc4. alsa.conf: # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion ## alias snd-card-0 snd-interwave ## alias snd-card-1 snd-ens1371 ## OSS/Free portion ## alias sound-slot-0 snd-card-0 ## alias sound-slot-1 snd-card-1 ## # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 ## alias sound-service-1-0 snd-mixer-oss ## alias sound-service-1-3 snd-pcm-oss ## alias sound-service-1-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 #options snd-hda-intel model=thinkpad options snd-hda-intel model=thinkpad enable=1 index=0 options snd-hda-intel model=hdmi enable=1 index=-2 lspci -vv and dmesg goes here: [1]http://pastebin.ca/2510939 Any Help? References 1. http://pastebin.ca/2510939

4 9

[alsa-devel] [PATCH 0/3] alsa-lib - improve surround 2.1 support
by David Henningsson 21 Feb '14

21 Feb '14

So, in order to support 2.1 surround on both 4-channels (e g FL FR LFE LFE, on internal subwoofer) and 6-channels (e g FL FR RL RR C LFE, on desktops with 5.1 out), I needed to make the route plugin more chmap aware. This also included allowing a new syntax for the ttable, which I think is quite nice. The new syntax also falls back to ALSAs standard chmap in case the hardware does not support it. David Henningsson (3): route: Allow chmap syntax for slave channels in ttable route: Select slave chmap based on ttable information conf: Allow 2.1 surround to use different number of channels src/conf/pcm/surround21.conf | 7 +- src/pcm/pcm_route.c | 304 ++++++++++++++++++++++++++++++++++++++----- 2 files changed, 271 insertions(+), 40 deletions(-) -- 1.9.0

2 6

[alsa-devel] [PATCH] rawmidi: remove undefined functions.
by Takashi Sakamoto 21 Feb '14

21 Feb '14

'snd_rawmidi_transmit_reset()' and 'snd_rawmidi_receive_reset()' are declared but not defined. This state has been continue over 10 years. So let us remove them. --- include/sound/rawmidi.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index adf0885..311dafe 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -157,10 +157,8 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, /* callbacks */ -void snd_rawmidi_receive_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count); -void snd_rawmidi_transmit_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count); -- 1.8.3.2

2 3

[alsa-devel] Enable more HP machine to enable mute led
by Kailang 21 Feb '14

21 Feb '14

Hi Takashi, HP want to enable mute led functions for more machine. I created patch as attach patch. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 000/102 - chunk#5] Add SOC_ENUM_SINGLE_CONST() & co
by Takashi Iwai 21 Feb '14

21 Feb '14

This is the last chunk of patch series, which introduces two new macros, SOC_ENUM_SINGLE_CONST() and SOC_ENUM_DOUBLE_CONST() in order to simplify the codes without ARRAY_SIZE() in the caller side. Takashi

2 30

Re: [alsa-devel] Headphone output problems with HP ProDesk 600 G1 (Realtek ALC221)
by Raymond Yau 20 Feb '14

20 Feb '14

>> I have a HP ProDesk 600 G1 machine which seems to have some strange behavior with the front headphone jack(s) running under Fedora 20 (running 3.13.2 kernel): >> This machine has two front jacks: a headphone jack and a microphone/headphone jack. When the headphones are plugged into the headphone jack, there's usually no change in the sound outputs list in the Sound control panel. The audio (mostly) works if the audio output device is set to Analog Output, but sometimes spuriously changes back to Speakers and the headphone output stops working (I suspect maybe from a spurious jack-sensing event, though I'm not certain). More confusingly, a couple of times I have gotten it into the state (through various fiddling with jacks and playing in the Sound control panel) where plugging into this jack is detected and causes the Headphone output to be selected in the control panel. I haven't been able to figure out consistently what causes this to happen. >> If the headphone is plugged into the microphone/headphone jack, then the control panel consistently switches to the Headphone output and consistently switches back the way it was when it's unplugged. However, there's no actual audio output coming out of that jack. (From some of the HP manuals, it sounds like with the Windows driver, when you plug into this jack, it's supposed to pop up a prompt asking you whether you want to use it as a microphone or headphones.) >> The other strange thing is that intermittently, on the headphone jack you'll start getting a bunch of loud static (clicking noises) and any actual audio playback becomes very faint and distorted. >> I'm guessing perhaps something isn't being initialized/setup properly in the codec, causing this kind of erratic behavior. The alsa-info output is uploaded at: http://www.alsa-project.org/db/?f=594e837b7af07ffbe3473cf807465bc81a587880 >> I tried using hda-analyzer but it fails on this machine with: >> ValueError: wrong proc file format (unknown dig1 bit 'KAE') >> Any suggestions for things to try to debug this? Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 it is a driver bug which assign PCM playback Switch to Line Out Jack try the patch https://bugzilla.kernel.org/attachment.cgi?id=124971 HP ProDesk 600 G1 SFF is an All-In-One pc , should headphone and line out share the volume control if headphone can mute the line out or speaker Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out Speaker' Item0: 'Line Out Speaker' Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC221 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x57 0x57] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM

2 8

Re: [alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by David Henningsson 20 Feb '14

20 Feb '14

On 02/19/2014 03:57 PM, Takashi Sakamoto wrote: > Hi all, > > I have two questions about the usage of this macro. > > Should I use this macro for new drivers which I'm working for?' > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… > > > My drivers support some firewire devices, which is desinged for > studio-use and have many channels. Its combination of channels are not > designed for surround sound like 5.1ch. > > Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. > But there are no drivers which use this macro in current tree, against > my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so > on) should use this macro. > > Are there any reasons? I've never heard of it before. I had a quick look in alsa-lib and couldn't find any difference in handling between CLASS_GENERIC and CLASS_MULTI. It's probably something historic that nobody uses any more. Maybe people who have been along longer knows what it is. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 3

[alsa-devel] [PATCH] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 20 Feb '14

20 Feb '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. 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 | 490 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 507 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 de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ 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..847e079 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,490 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#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); +static unsigned int devices_used[SNDRV_CARDS/sizeof(int)]; +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 len) +{ + unsigned int 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, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, 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, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command 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(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__, 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, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + 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(dev, "unable to submit urb. OOM!?\n"); +} + +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 (!(devices_used[card_index/sizeof(int)*8] & + (1 << card_index % (sizeof(int)*8)))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(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); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + 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); + devices_used[card_index/sizeof(int)*8] |= + 1 << (card_index%(sizeof(int)*8)); + + 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); + + devices_used[bcd2k->card_index/sizeof(int)*8] &= + ~(1 << (bcd2k->card_index%(sizeof(int)*8))); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .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

2 1

[alsa-devel] [PATCH v4 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 20 Feb '14

20 Feb '14

Hi, These are the remaining patches from v3 of this version. They add master clock and regulator support. In v4 I added a check for the regulator dependencies and updated the binding documentation for them. I fixed the master clock patch to shutdown the master clock after the PLL and added another patch to shutdown all dividers before the PLL. Regards, Markus Changes in v4: - Fix clock shutdown order - Add regulator dependency checks - Revert the previously added tlv320aic32x4 compatible from the binding documentation Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Revert "ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible" .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 126 ++++++++++++++++++--- 3 files changed, 123 insertions(+), 16 deletions(-) -- 1.8.5.3

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[alsa-devel] Need Help: Repeated Jackhammer noise after several hours of play and heavy cpu usage in snd_pcm_dmix_sync_area
by Adarsh 20 Feb '14

20 Feb '14

Hi, We are facing a issue where if gdb is attached to the our long running sound process and continued, jackhammer sound is played over and over again abusing the speakers. Process details: 1. We are using RW_INTERLEAVED access as described in write_loop method in /test/pcm.c and opening 'default' device. 2. However to disable the underruns, the snd_pcm_sw_params_set_stop_threshold is set to 0x7fffffff which is greater than boundary value. 3.Period size is 1024 and buffer size is 2048. We try to write period_size no of frames using snd_pcm_writei. (I am attaching a sample program which shows this behaviour.) Steps to Reproduce: 1. Left process running for over 12 hours. 2. gdb is attached to the process and continued inside gdb. 3. At this point, the thread went into a tight loop taking >100% cpu and repeatedly playing same sound. It did not recover for a long time. Stack trace shows that 'snd_pcm_dmix_sync_area' in pcm_dmix.c is taking a very long time to return because the 'size' is a large unsinged interger and mix_areas is getting called with 'transfer' value of <=2048. If snd_pcm_sw_params_set_stop_threshold is not set or when application is running for shorter duration, attaching gdb does not show this behaviour. Is it that setting of stop threshold to 0x7fffffff is triggering this and attaching gdb is making the buffer pointers misbehave? If so, what can be done if we do not want PCM to stop? I have tried by setting silence threshold and silence size to let alsa play silence by itself but still observed the issue. Below is the stack trace and PCM dump: (gdb) bt #0 0xb7c44463 in mix_areas_32_smp (size=1360, dst=0xb6855000, src=0x84a7510, sum=0xb6845000, dst_step=4, src_step=4, sum_step=4) at pcm_dmix_i386.h:261 #1 0xb7c44d2b in mix_areas (dmix=0x84a7110, src_areas=0x84a51d0, dst_areas=0x84a6f00, src_ofs=0, dst_ofs=7168, size=1024) at pcm_dmix.c:193 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 #3 0xb7c46680 in snd_pcm_dmix_mmap_commit (pcm=0x84a71e0, offset=0, size=1024) at pcm_dmix.c:813 #4 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a71e0, offset=0, frames=1024) at pcm.c:6501 #5 0xb7c1804e in snd_pcm_plugin_mmap_commit (pcm=0x84a6d18, offset=0, size=1024) at pcm_plugin.c:434 #6 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a6d18, offset=0, frames=1024) at pcm.c:6501 #7 0xb7c17956 in snd_pcm_plugin_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024) at pcm_plugin.c:290 #8 0xb7c074bb in snd1_pcm_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024, func=0xb7c177fc <snd_pcm_plugin_write_areas>) at pcm.c:6667 #9 0xb7c17cd5 in snd_pcm_plugin_writei (pcm=0x84a5390, buffer=0x84ab518, size=1024) at pcm_plugin.c:361 #10 0xb7bff1a7 in _snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm_local.h:516 #11 0xb7bff16c in snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm.c:1258 (gdb) f 2 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 371 pcm_dmix.c: No such file or directory. (gdb) info locals dmix = 0x84a7110 slave_hw_ptr = 234156032 slave_appl_ptr = 7168 slave_size = 1057171456 appl_ptr = 0 size = 1018295296 transfer = 1024 src_areas = 0x84a51d0 dst_areas = 0x84a6f00 (gdb) p *dmix $1 = {type = SND_PCM_TYPE_DMIX, ipc_key = 5678293, ipc_perm = 432, ipc_gid = 17, semid = 32768, shmid = 65536, shmptr = 0xb6859000, spcm = 0x84a7340, appl_ptr = 217121792, last_appl_ptr = 233882624, hw_ptr = 233875428, avail_max = 0, slave_appl_ptr = 234156032, slave_hw_ptr = 234147996, slave_period_size = 1024, slave_buffer_size = 8192, slave_boundary = 1073741824, sync_ptr = 0xb7c4557c <snd_pcm_dmix_sync_ptr>, state = SND_PCM_STATE_RUNNING, trigger_tstamp = {tv_sec = 1392384066, tv_nsec = 8308000}, server = 0, client = 0, comm_fd = 0, hw_fd = 4, timer_fd = { fd = 3, events = 41, revents = 0}, poll_fd = 3, tread = 1, timer_need_poll = 0, timer_event_suspend = 131072, timer_event_resume = 262144, server_fd = 0, server_pid = 0, timer = 0x8496b90, interleaved = 1, slowptr = 1, max_periods = 0, channels = 2, bindings = 0x0, u = {dmix = { shmid_sum = 98305, sum_buffer = 0xb6837000, mix_areas_16 = 0xb7c443a8 <mix_areas_16_smp_mmx>, mix_areas_32 = 0xb7c44415 <mix_areas_32_smp>, mix_areas_24 = 0xb7c4451b <mix_areas_24_smp_cmov>, mix_areas_u8 = 0, remix_areas_16 = 0xb7c44638 <remix_areas_16_smp_mmx>, remix_areas_32 = 0xb7c446a5 <remix_areas_32_smp>, remix_areas_24 = 0xb7c447ab <remix_areas_24_smp_cmov>, remix_areas_u8 = 0}, dsnoop = {<No data fields>}, dshare = {chn_mask = 13151478482154389505}}, server_free = 0} (gdb) p *pcm $2 = {dl_handle = 0x0, name = 0x84a6e48 "dmix:0", type = SND_PCM_TYPE_DMIX, stream = SND_PCM_STREAM_PLAYBACK, mode = 0, poll_fd_count = 1, poll_fd = 3, poll_events = 1, setup = -1, monotonic = 0, access = SND_PCM_ACCESS_MMAP_INTERLEAVED, format = SND_PCM_FORMAT_S32_LE, subformat = SND_PCM_SUBFORMAT_STD, channels = 2, rate = 44100, period_size = 1024, period_time = 23219, periods = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, tstamp_mode = SND_PCM_TSTAMP_NONE, period_step = 1, avail_min = 1024, period_event = 0, start_threshold = 1024, stop_threshold = 2147483647, silence_threshold = 0, silence_size = 0, boundary = 1073741824, info = 2151743747, msbits = 32, rate_num = 44100, rate_den = 1, hw_flags = 0, fifo_size = 0, buffer_size = 2048, buffer_time = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, sample_bits = 32, frame_bits = 64, appl = {master = 0x0, ptr = 0x84a7130, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, hw = { master = 0x0, ptr = 0x84a7138, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, min_align = 1, mmap_rw = 1, mmap_shadow = 0, donot_close = 0, mmap_channels = 0x84a54a0, running_areas = 0x84a51d0, stopped_areas = 0x0, ops = 0xb7c91860, fast_ops = 0xb7c918a0, op_arg = 0x84a71e0, fast_op_arg = 0x84a71e0, private_data = 0x84a7110, async_handlers = {next = 0x84a72e4, prev = 0x84a72e4}} PCM dump: Plug PCM: Linear conversion PCM (S32_LE) 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 : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8192 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 Thanks

3 3

[alsa-devel] [PATCH 00/17 v2] Update locking for snd_soc_dapm_xxxx_pin functions
by Charles Keepax 20 Feb '14

20 Feb '14

This patchset updates the locking around the snd_soc_dapm_xxxx_pin functions. First we add an unlocked version of the functions and add locking to the standard call, this also requires us to remove the few places where these functions are externally locked. Then we patch usages of the functions that look like they are expecting to do an atomic update to do so, using external locking. Thanks, Charles Updates since v1: - Added helper functions to lock/unlock the dapm mutex - Added an unlocked version of snd_soc_dapm_sync, so it can be called under the same mutex lock as the pin functions, this is not required just saves a mutex lock Charles Keepax (17): Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ---- include/sound/soc-dapm.h | 9 ++ include/sound/soc.h | 9 ++ sound/soc/codecs/adav80x.c | 17 +++-- sound/soc/codecs/wm5100.c | 12 ++- sound/soc/codecs/wm8962.c | 13 ++- sound/soc/codecs/wm8994.c | 45 ++++++---- sound/soc/codecs/wm8996.c | 9 ++- sound/soc/intel/mfld_machine.c | 65 +++++++++----- sound/soc/omap/ams-delta.c | 45 ++++++---- sound/soc/omap/n810.c | 22 +++-- sound/soc/omap/rx51.c | 22 +++-- sound/soc/pxa/corgi.c | 42 +++++---- sound/soc/pxa/magician.c | 22 +++-- sound/soc/pxa/spitz.c | 51 ++++++----- sound/soc/pxa/tosa.c | 28 ++++--- sound/soc/soc-dapm.c | 160 ++++++++++++++++++++++++++++++--- 18 files changed, 399 insertions(+), 203 deletions(-) -- 1.7.2.5

4 39

[alsa-devel] [PATCH] ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform
by Jarkko Nikula 20 Feb '14

20 Feb '14

Initial implementation of this driver focused only matching SST ACPI ID with single machine driver and same firmware file per platform. It was known restriction to be improved incrementally. This patch is now changing this that SST ACPI ID refers purely to platform specific data which refers to machine drivers on this platform, not vice versa. Matching machine driver is found by looking at ACPI ID which would best match with the driver. Typically this would be the ACPI ID of audio codec but is not tied to it. This patch also changes that DSP firmware name is machine not platform specific. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This goes on top of my "[PATCHv2] ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing". http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072812.h… --- sound/soc/intel/sst-acpi.c | 84 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 58 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 64154a77d904..c7e36c9ee52f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -26,9 +26,20 @@ #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 #define SST_LPT_DSP_DMA_SIZE (1024 - 1) +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + /* ACPI ID for the matching machine driver. Audio codec for instance */ + const u8 id[ACPI_ID_LEN]; + /* machine driver name */ + const char *drv_name; + /* firmware file name */ + const char *fw_filename; +}; + /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; + struct sst_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -37,24 +48,17 @@ struct sst_acpi_desc { int resindex_dma_base; /* Unique number identifying the SST core on platform */ int sst_id; - /* firmware file name */ - const char *fw_filename; /* DMA only valid when resindex_dma_base != -1*/ int dma_engine; int dma_size; }; -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - const char *drv_name; - struct sst_acpi_desc *res_desc; -}; - struct sst_acpi_priv { struct platform_device *pdev_mach; struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; + struct sst_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -64,10 +68,11 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; + struct sst_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { - dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename); return; } @@ -83,6 +88,28 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) return; } +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_acpi_mach *sst_acpi_find_machine( + struct sst_acpi_mach *machines) +{ + struct sst_acpi_mach *mach; + bool found = false; + + for (mach = machines; mach->id[0]; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -102,8 +129,13 @@ static int sst_acpi_probe(struct platform_device *pdev) if (!id) return -ENODEV; - mach = (struct sst_acpi_mach *)id->driver_data; - desc = mach->res_desc; + desc = (struct sst_acpi_desc *)id->driver_data; + mach = sst_acpi_find_machine(desc->machines); + if (mach == NULL) { + dev_err(dev, "No matching ASoC machine driver found\n"); + return -ENODEV; + } + sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; @@ -154,7 +186,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(sst_acpi->pdev_mach); /* continue SST probing after firmware is loaded */ - ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename, dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); if (ret) platform_device_unregister(sst_acpi->pdev_mach); @@ -175,45 +207,45 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } +static struct sst_acpi_mach haswell_machines[] = { + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_haswell_desc = { .drv_name = "haswell-pcm-audio", + .machines = haswell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_LYNX_POINT, - .fw_filename = "intel/IntcSST1.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; +static struct sst_acpi_mach broadwell_machines[] = { + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_broadwell_desc = { .drv_name = "haswell-pcm-audio", + .machines = broadwell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_WILDCAT_POINT, - .fw_filename = "intel/IntcSST2.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach haswell_mach = { - .drv_name = "haswell-audio", - .res_desc = &sst_acpi_haswell_desc, -}; - -static struct sst_acpi_mach broadwell_mach = { - .drv_name = "broadwell-audio", - .res_desc = &sst_acpi_broadwell_desc, -}; - static struct acpi_device_id sst_acpi_match[] = { - { "INT33C8", (unsigned long)&haswell_mach }, - { "INT3438", (unsigned long)&broadwell_mach }, + { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, + { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- 1.8.5.3

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[alsa-devel] [PATCH] ALSA: Export snd_pcm_constraint_mask64()
by Mark Brown 20 Feb '14

20 Feb '14

From: Mark Brown <broonie(a)linaro.org> Allow modules to use it, fixing a build failure when the newly added ADAU1977 driver is built as a module. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- OK to apply this to the ASoC tree? sound/core/pcm_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6630a4e..ce83def 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1240,6 +1240,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval -- 1.9.0.rc3

2 2

[alsa-devel] [PATCH] ALSA: hda - Enable front audio jacks on one HP desktop model
by Hui Wang 20 Feb '14

20 Feb '14

The front headphone and mic jackes on a HP desktop model (Vendor Id: 0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this machine has 8 physical ports, 6 of them are routed to rear jackes and all of them work very well, while the remaining 2 ports are routed to front headphone and mic jackes, but the corresponding pin complex node are not defined correctly. After apply this fix, the front audio jackes can work very well. BugLink: https://bugs.launchpad.net/bugs/1282369 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Gerald Yang <gerald.yang(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7311bad..d75f90a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,7 +83,8 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, - STAC_92HD73XX_MODELS + STAC_92HD73XX_MODELS, + STAC_92HD89XX_HP_FRONT_JACK }; enum { @@ -1795,6 +1796,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1913,6 +1920,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1973,6 +1984,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; -- 1.8.1.2

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[alsa-devel] [PATCH 000/102 - chunk#2] Add const to SOC_ENUM_*_DECL() macros
by Takashi Iwai 20 Feb '14

20 Feb '14

The first patch adds the const modifier to SOC_ENUM_SINGLE_DECL() & co, and the rest are to remove superfluous const modifiers. Takashi

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[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 20 Feb '14

20 Feb '14

The following changes since commit 6d0abeca3242a88cab8232e4acd7e2bf088f3bc2: Linux 3.14-rc3 (2014-02-16 13:30:25 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc3 for you to fetch changes up to 895be5b31e5175bef575008aadb4f0a27b850daa: Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus (2014-02-19 13:13:58 +0900) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few fixes, all driver speccific ones. The DaVinci ones aren't as clear as they should be from the subject lines on the commits but they fix issues which will prevent correct operation in some use cases and only affect that particular driver so are reasonably safe. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da9055: Fix device registration of PMIC and CODEC devices Alexander Shiyan (1): ASoC: txx9aclc_ac97: Fix kernel crash on probe Dylan Reid (1): ASoC: max98090: sync regcache on entering STANDBY Jarkko Nikula (1): ASoC: rt5640: Add ACPI ID for Intel Baytrail Lars-Peter Clausen (1): ASoC: blackfin: Fix machine driver Kconfig dependencies Mark Brown (2): Merge remote-tracking branches 'asoc/fix/blackfin', 'asoc/fix/da9055', 'asoc/fix/davinci', 'asoc/fix/fsl', 'asoc/fix/fsl-esai', 'asoc/fix/max98090', 'asoc/fix/rt5640', 'asoc/fix/samsung' and 'asoc/fix/txx9aclc-ac97' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus Peter Ujfalusi (4): ASoC: davinci-mcasp: Harmonize the sub hw_params function names ASoC: davinci-mcasp: Configure xxTDM, xxFMT and xxFMCT registers synchronously ASoC: davinci-mcasp: Consolidate pm_runtime_get/put() use in the driver ASoC: davinci-evm: Add pm callbacks to platform driver Sachin Kamat (2): ASoC: samsung: Remove invalid dependencies ASoC: samsung: Fix trivial typo Shawn Guo (1): ASoC: fsl: fix pm support of machine drivers Stephen Warren (1): ASoC: max98090: make REVISION_ID readable Wei Yongjun (1): ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data Xiubo Li (1): ASoC: fsl-esai: fix ESAI TDM slot setting drivers/mfd/da9055-i2c.c | 12 +++++- sound/soc/blackfin/Kconfig | 11 +++--- sound/soc/codecs/da9055.c | 11 +++++- sound/soc/codecs/max98090.c | 21 +++++----- sound/soc/codecs/rt5640.c | 1 + sound/soc/codecs/wm8993.c | 1 - sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-mcasp.c | 83 ++++++++++++++++++--------------------- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-sgtl5000.c | 10 +++-- sound/soc/fsl/imx-wm8962.c | 11 ++++-- sound/soc/samsung/Kconfig | 6 +-- sound/soc/txx9/txx9aclc-ac97.c | 8 ++-- 15 files changed, 99 insertions(+), 84 deletions(-)

2 1

[alsa-devel] Need your advice: Add a new communication inteface between HD-Audio and Gfx drivers for hotplug notification/ELD update
by Lin, Mengdong 20 Feb '14

20 Feb '14

Dear audio and gfx stakeholders, We hope to add a new interface between audio and gfx driver, for gfx driver to notify audio about HDMI/DP hot-plug and ELD update. Would you please share some comments on the proposal below? Background of this issue: On Intel Haswell/Broadwell platforms, there is a HW restriction that after the display HD-Audio controller is in D3, it cannot be waken up by HDMI/DP hot-plug. Consequently, although the gfx driver can still detect the HDMI/DP hot-plug, audio driver has no idea about this and cannot notify user space whether the external HDMI/DP monitor is available for audio playback, because the audio controller cannot wake up to D0 and receive HW unsolicited event about hot-plug from the audio codec. This limitation will affect user space to decide whether we can output audio over HDMI/DP. To solve the above limitation, Takashi suggested to add a new communication interface between audio and gfx driver: create a common object containing the ops registered by both graphics and audio drivers, then communicate through it, something like vga_switcheroo. Is it okay to create this kernel object in i915 driver? I915 can export an API like "display_register_audio_client" for audio driver to register a client and hot-plug notification ops. I915 can also call some API like "display_register_gfx_client" itself and register ops for audio driver to query monitor presence and ELD info on a specific port. It would be faster for audio driver than quering ELD by command/response over the HD-A bus, thus avoid delay in i915 mode set. This will also avoid waking up the audio devices unnecessarily if the user space does not really want to use HDMI/DP for audio playback. Whenever i195 enables/disables audio on a port in modeset, it can call some API like "display_set_audio_state()" on this kernel object and trigger notifications to the audio driver. When the audio driver is probed (in the delayed probe stage), it can request i915 API symbol to register the audio client for this communication kernel object. Since the 1st i915 mode set may happen before audio driver registers the ops, we'll let audio driver check ELD once after registering the audio client ops. And for the platforms which uses this communication interface, we can disable unsolicited event for HDMI/DP hot-plug in the audio driver. We hope to hear your feedback and start to work out more details. Thanks & Best Regards Mengdong

8 21

[alsa-devel] [PATCH 000/105] Rename soc_enum.max field
by Takashi Iwai 20 Feb '14

20 Feb '14

Hi, this is the first chunk of the whole patch serires, preliminary works for renaming the confusing field name and minor clean ups in the relevant places. After this chunk, SOC_ENUM_*_DECL() macro cleanup and the actual fixes for the wrong number of enum items will follow. Then introduce SOC_ENUM_SINGLE_CONST() & co for omitting the explicit ARRAY_SIZE() call, and apply it in the later patches. Takashi

6 11

[alsa-devel] [PATCH 000/102 - chunk#3] Actually fix wrong enum items
by Takashi Iwai 20 Feb '14

20 Feb '14

Now, this patch chunk fixes the actual bugs in various codec drivers that have either off-by-one number of items or lead to array overflow. sta32x is the latter one, and this can be picked up for 3.14, too. The other off-by-one fixes would change the driver behavior (now user will see more items), thus they are potential regression risks although the "fixes" are correct, per se. Takashi

5 28

[alsa-devel] [PATCH 1/2] ASoC: alc5623: Convert to direct regmap API usage
by Mark Brown 20 Feb '14

20 Feb '14

From: Mark Brown <broonie(a)linaro.org> 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> --- sound/soc/codecs/alc5623.c | 87 ++++++++++++++++++++++++++-------------------- 1 file changed, 49 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d303628..36cf7e8 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -38,26 +39,13 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u8 id; unsigned int sysclk; - u16 reg_cache[ALC5623_VENDOR_ID2+2]; unsigned int add_ctrl; unsigned int jack_det_ctrl; }; -static void alc5623_fill_cache(struct snd_soc_codec *codec) -{ - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; - - /* not really efficient ... */ - codec->cache_bypass = 1; - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = snd_soc_read(codec, i); - codec->cache_bypass = 0; -} - static inline int alc5623_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, ALC5623_RESET, 0); @@ -869,18 +857,28 @@ static struct snd_soc_dai_driver alc5623_dai = { static int alc5623_suspend(struct snd_soc_codec *codec) { + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(alc5623->regmap, true); + return 0; } static int alc5623_resume(struct snd_soc_codec *codec) { - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; /* Sync reg_cache with the hardware */ - for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) - snd_soc_write(codec, i, cache[i]); + regcache_cache_only(alc5623->regmap, false); + ret = regcache_sync(alc5623->regmap); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync register cache: %d\n", + ret); + regcache_cache_only(alc5623->regmap, true); + return ret; + } alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -900,14 +898,14 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + codec->control_data = alc5623->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; } alc5623_reset(codec); - alc5623_fill_cache(codec); /* power on device */ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -980,9 +978,15 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, - .reg_cache_size = ALC5623_VENDOR_ID2+2, - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, +}; + +static const struct regmap_config alc5623_regmap = { + .reg_bits = 8, + .val_bits = 16, + .reg_stride = 2, + + .max_register = ALC5623_VENDOR_ID2, + .cache_type = REGCACHE_RBTREE, }; /* @@ -996,19 +1000,32 @@ static int alc5623_i2c_probe(struct i2c_client *client, { struct alc5623_platform_data *pdata; struct alc5623_priv *alc5623; - int ret, vid1, vid2; + unsigned int vid1, vid2; + int ret; - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); - if (vid1 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); + if (alc5623 == NULL) + return -ENOMEM; + + alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap); + if (IS_ERR(alc5623->regmap)) { + ret = PTR_ERR(alc5623->regmap); + dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret); + return ret; + } + + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); + return ret; } vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); - 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; } if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { @@ -1021,11 +1038,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), - GFP_KERNEL); - if (alc5623 == NULL) - return -ENOMEM; - pdata = client->dev.platform_data; if (pdata) { alc5623->add_ctrl = pdata->add_ctrl; @@ -1048,7 +1060,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); -- 1.9.0.rc3

1 1

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCHv2] ASoC: Intel: Fix build for sst-dsp.c on PPC architecture
by Liam Girdwood 19 Feb '14

19 Feb '14

Disable build on non X86 architectures except for compile testing. This fixes the following build errors on PPC and adds an option for testing the build on other architectures as suggested by Mark Brown :- sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_write': sound/soc/intel/sst-dsp.c:218:2: error: implicit declaration of function 'memcpy_toio' [-Werror=implicit-function-declaration] memcpy_toio(sst->mailbox.out_base, message, bytes); ^ sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_read': sound/soc/intel/sst-dsp.c:231:2: error: implicit declaration of function 'memcpy_fromio' [-Werror=implicit-function-declaration] memcpy_fromio(message, sst->mailbox.out_base, bytes); ^ Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/sst-dsp.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c962432..4bb802b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,6 +15,7 @@ config SND_SST_MFLD_PLATFORM config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI + depends on (X86 || COMPILE_TEST) help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index e0ad2e5..6e22c12 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -19,6 +19,7 @@ #include <linux/interrupt.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/io.h> #include "sst-dsp.h" #include "sst-dsp-priv.h" -- 1.8.3.2

2 1

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCH] ASoC: cs4271: claim reset GPIO in bus probe function
by Daniel Mack 19 Feb '14

19 Feb '14

Move the GPIO acquisition from the codec to the bus probe functions. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 60 +++++++++++++++++++++++++++++++---------------- 1 file changed, 40 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd9..6bfe021 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -540,14 +540,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; - int gpio_nreset = -EINVAL; bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { - gpio_nreset = of_get_named_gpio(codec->dev->of_node, - "reset-gpio", 0); - if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; @@ -559,27 +555,19 @@ static int cs4271_probe(struct snd_soc_codec *codec) #endif if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_nreset >= 0) - if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) - gpio_nreset = -EINVAL; - if (gpio_nreset >= 0) { + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ - gpio_direction_output(gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); udelay(1); - gpio_set_value(gpio_nreset, 1); + gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ udelay(1); } - cs4271->gpio_nreset = gpio_nreset; - ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); @@ -641,14 +629,45 @@ static const struct regmap_config cs4271_spi_regmap = { .volatile_reg = cs4271_volatile_reg, }; -static int cs4271_spi_probe(struct spi_device *spi) +static int cs4271_common_probe(struct device *dev, + struct cs4271_private **c) { + struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; - cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) return -ENOMEM; + if (of_match_device(cs4271_dt_ids, dev)) + cs4271->gpio_nreset = + of_get_named_gpio(dev->of_node, "reset-gpio", 0); + + if (cs4271plat) + cs4271->gpio_nreset = cs4271plat->gpio_nreset; + + if (gpio_is_valid(cs4271->gpio_nreset)) { + int ret; + + ret = devm_gpio_request(dev, cs4271->gpio_nreset, + "CS4271 Reset"); + if (ret < 0) + return ret; + } + + *c = cs4271; + return 0; +} + +static int cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + int ret; + + ret = cs4271_common_probe(&spi->dev, &cs4271); + if (ret < 0) + return ret; + spi_set_drvdata(spi, cs4271); cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); if (IS_ERR(cs4271->regmap)) @@ -698,10 +717,11 @@ static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&client->dev, &cs4271); + if (ret < 0) + return ret; i2c_set_clientdata(client, cs4271); cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); -- 1.8.5.3

4 5

[alsa-devel] [PATCH v4] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 19 Feb '14

19 Feb '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. 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 | 488 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 505 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 de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ 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..6965a89 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,488 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#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 bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +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 len) +{ + unsigned int 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, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, 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, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command 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(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__, 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, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + 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(dev, "unable to submit urb. OOM!?\n"); +} + +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 (enable[card_index] && !(devices_used & (1 << card_index))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(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); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + 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); + devices_used |= 1 << card_index; + + 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); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .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

3 2

[alsa-devel] [PATCHv2] ASoC: Intel: Fix sparse warnings for firmware loader.
by Liam Girdwood 19 Feb '14

19 Feb '14

Changes since V1 :- o Fixed sparse warning for block_module_remove o Fixed additional sparse warnings when building entire driver. Sparse gives us the following warnings on sst-firmware.c CHECK sound/soc/intel/sst-firmware.c sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:39:34: expected void volatile [noderef] <asn:2>*dst sound/soc/intel/sst-firmware.c:39:34: got void * sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:417:36: expected void *dest sound/soc/intel/sst-firmware.c:417:36: got void [noderef] <asn:2>* sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static? and CC [M] sound/soc/intel/sst-dsp.o sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* This patch removes these warnings Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 7 +++++-- sound/soc/intel/sst-firmware.c | 2 +- 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index 6135564..fe8e81a 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -41,8 +41,10 @@ struct sst_ops { u64 (*read64)(void __iomem *addr, u32 offset); /* DSP I/DRAM IO */ - void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); - void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src, + size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src, + size_t bytes); void (*dump)(struct sst_dsp *); @@ -296,6 +298,7 @@ struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); +int sst_block_module_remove(struct sst_module *module); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 31cd154..dee7eb5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void sst_memcpy32(void *dest, void *src, u32 bytes) +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; -- 1.8.3.2

3 9

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCHv2] ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing
by Jarkko Nikula 19 Feb '14

19 Feb '14

We originally thought to request SST audio DSP firmware during the SST platform driver initialization. However plain request_firmware doesn't work in driver probe paths if userspace is not ready to handle it. For instance when drivers are built-in. Implementing asynchronous firmware request in SST platform driver initialization complicates code needlessly since it anyway will fail if firmware is missing. This is more simple to handle by requesting firmware asynchronously in sst_acpi_probe() and register SST platform only after firmware is loaded. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- V2: - Fix leak in sst_acpi_probe that sst_acpi->pdev_mach wasn't unregistered in case of request_firmware_nowait fails - Test is sst_acpi->pdev_pcm valid before unregistering it in sst_acpi_remove. It can contain not only NULL or valid pointer but an error code too --- sound/soc/intel/sst-acpi.c | 56 ++++++++++++++++++++++++++++++++-------------- sound/soc/intel/sst-dsp.h | 2 +- 2 files changed, 40 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index aba73ca8a923..64154a77d904 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -16,6 +16,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/firmware.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -56,6 +57,32 @@ struct sst_acpi_priv { struct sst_acpi_desc *desc; }; +static void sst_acpi_fw_cb(const struct firmware *fw, void *context) +{ + struct platform_device *pdev = context; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; + struct sst_acpi_desc *desc = sst_acpi->desc; + + sst_pdata->fw = fw; + if (!fw) { + dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + return; + } + + /* register PCM and DAI driver */ + sst_acpi->pdev_pcm = + platform_device_register_data(dev, desc->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_pcm)) { + dev_err(dev, "Cannot register device %s. Error %d\n", + desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm)); + } + + return; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -79,7 +106,6 @@ static int sst_acpi_probe(struct platform_device *pdev) desc = mach->res_desc; sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; - sst_pdata->fw_filename = desc->fw_filename; sst_acpi->desc = desc; if (desc->resindex_dma_base >= 0) { @@ -118,37 +144,33 @@ static int sst_acpi_probe(struct platform_device *pdev) } } - /* register PCM and DAI driver */ - sst_acpi->pdev_pcm = - platform_device_register_data(dev, desc->drv_name, -1, - sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_pcm)) - return PTR_ERR(sst_acpi->pdev_pcm); - - /* register machine driver */ platform_set_drvdata(pdev, sst_acpi); + /* register machine driver */ sst_acpi->pdev_mach = platform_device_register_data(dev, mach->drv_name, -1, sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_mach)) { - ret = PTR_ERR(sst_acpi->pdev_mach); - goto sst_err; - } + if (IS_ERR(sst_acpi->pdev_mach)) + return PTR_ERR(sst_acpi->pdev_mach); - return ret; + /* continue SST probing after firmware is loaded */ + ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); + if (ret) + platform_device_unregister(sst_acpi->pdev_mach); -sst_err: - platform_device_unregister(sst_acpi->pdev_pcm); return ret; } static int sst_acpi_remove(struct platform_device *pdev) { struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; platform_device_unregister(sst_acpi->pdev_mach); - platform_device_unregister(sst_acpi->pdev_pcm); + if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm)) + platform_device_unregister(sst_acpi->pdev_pcm); + release_firmware(sst_pdata->fw); return 0; } diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index d134359fecac..3730fd324455 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -152,7 +152,7 @@ struct sst_pdata { int irq; /* Firmware */ - const char *fw_filename; + const struct firmware *fw; /* DMA */ u32 dma_base; -- 1.8.5.3

3 3

[alsa-devel] [PATCH] ASoC: cs4271: free reset gpio in cs4271_remove()
by Daniel Mack 19 Feb '14

19 Feb '14

The reset GPIO line is acquired in cs4271_probe(), so it should be freed from cs4271_remove() as well. This way, the driver can unloaded and reloaded more than once. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd9..aeaa5c6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -604,9 +604,11 @@ static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(cs4271->gpio_nreset)) + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + devm_gpio_free(codec->dev, cs4271->gpio_nreset); + } return 0; }; -- 1.8.5.3

4 6

[alsa-devel] [PATCH] ALSA: core: Fix missing card sysfs contents
by Takashi Iwai 19 Feb '14

19 Feb '14

While moving the card device into struct snd_card, the reference to the assigned card in sysfs show/store callbacks were forgotten to be refreshed, still accessing to the no longer used drvdata. Fix these places to refer correctly via container_of(). Also, remove the superfluous NULL checks since it's guaranteed to be non-NULL now. Fixes: 8bfb181c17d2 ('ALSA: Embed card device into struct snd_card') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 0095a530ff12..b7085eb19079 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -624,15 +624,15 @@ static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = dev_get_drvdata(dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)"); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return snprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t card_id_store_attr(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_card *card = dev_get_drvdata(dev); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); char buf1[sizeof(card->id)]; size_t copy = count > sizeof(card->id) - 1 ? sizeof(card->id) - 1 : count; @@ -664,8 +664,8 @@ static ssize_t card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = dev_get_drvdata(dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return snprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fix sparse warnings for firmware loader.
by Liam Girdwood 19 Feb '14

19 Feb '14

Sparse gives us the following warning on sst-firmware.c CHECK sound/soc/intel/sst-firmware.c sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:39:34: expected void volatile [noderef] <asn:2>*dst sound/soc/intel/sst-firmware.c:39:34: got void * sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:417:36: expected void *dest sound/soc/intel/sst-firmware.c:417:36: got void [noderef] <asn:2>* sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static? This patch removes this warning. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 31cd154..dee7eb5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void sst_memcpy32(void *dest, void *src, u32 bytes) +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ALSA: hda - Add QEMU codec vendor ID
by Takashi Iwai 19 Feb '14

19 Feb '14

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4be322b0d745..115502783b64 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -67,6 +67,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, { 0x8086, "Intel" }, { 0x8384, "SigmaTel" }, -- 1.8.5.2

1 0

[alsa-devel] snd_intel8x0_inside_vm()
by H. Peter Anvin 19 Feb '14

19 Feb '14

I just stumbled upon the function snd_intel8x0_inside_vm(), and quite frankly the code seems bizarre: /* detect KVM and Parallels virtual environments */ result = kvm_para_available(); #ifdef X86_FEATURE_HYPERVISOR result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); #endif if (!result) goto fini; Why don't we simply rely on the PCI SSIDs? -hpa

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/ca0132 - setup/cleanup streams
by Hsin-Yu Chao 19 Feb '14

19 Feb '14

When a HDMI stream is opened with the same stream tag as a following opened stream to ca0132, audio will be heard from two ports simultaneously. Fix this issue by change to use snd_hda_codec_setup_stream and snd_hda_codec_cleanup_stream instead, so that an inactive stream can be marked as 'dirty' when found with a conflict stream tag, and then get purified. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Reviewed-by: Chih-Chung Chang <chihchung(a)chromium.org> --- sound/pci/hda/patch_ca0132.c | 68 ++++++-------------------------------------- 1 file changed, 9 insertions(+), 59 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d1479..59104dc 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2662,60 +2662,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * PCM stuffs - */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) -{ - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd( - "ca0132_setup_stream: NID=0x%x, stream=0x%x, " - "channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - - if (!nid) - return; - - snd_printdd(KERN_INFO "ca0132_cleanup_stream: NID=0x%x\n", nid); - - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!val) - return; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -} - -/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2726,7 +2672,9 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + ca0132_toggle_dac_format(codec); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); return 0; } @@ -2745,7 +2693,7 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) msleep(50); - ca0132_cleanup_stream(codec, spec->dacs[0]); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); return 0; } @@ -2824,8 +2772,8 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); return 0; } @@ -2839,7 +2787,7 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - ca0132_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -4742,6 +4690,8 @@ static int patch_ca0132(struct hda_codec *codec) return err; codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; return 0; } -- 1.9.0.rc1.175.g0b1dcb5

2 7

[alsa-devel] [PATCH 1/2] ASoC: Intel: Move extended fw base and size fields in struct sst_pdata
by Jarkko Nikula 19 Feb '14

19 Feb '14

Move fw_base and fw_size fields in struct sst_pdata under ACPI data for clarifying that these are not related to firmware file but for platform specific extended firmware area reserved by the BIOS. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 0ce5c8d91794..d134359fecac 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -147,12 +147,12 @@ struct sst_pdata { u32 lpe_size; u32 pcicfg_base; u32 pcicfg_size; + u32 fw_base; + u32 fw_size; int irq; /* Firmware */ const char *fw_filename; - u32 fw_base; - u32 fw_size; /* DMA */ u32 dma_base; -- 1.8.5.3

3 7

[alsa-devel] [PATCH] ASoC: Add ADAU1977 CODEC driver
by Lars-Peter Clausen 19 Feb '14

19 Feb '14

This patch adds support for the ADAU1977, ADAU1978 and ADAU1979 audio CODEC devices. They are a family of 4-channel differential input audio ADC devices. They can be connected to either a SPI or I2C bus. The driver is implemented in three modules, one main module (adau1977.ko) which implements the device logic and one module each for SPI (adau1977-spi.ko) and I2C (adau1977-i2c.ko) bus access. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Split out devicetree support into a separate patch. Will be submitted after this has been merged. * gpiod_set_value -> gpiod_set_value_cansleep * Split adau1977_set_power() into adau1977_disable_power() and adau1977_enable_power() as most of the code for those two cases was on different paths and was a bit hard to follow due to both being implemented in the same function. * Moved duplicated rate to FS field setting lookup to separate helper function. * Drop 'if (stream == SNDRV_PCM_STREAM_PLAYBACK) ...' in adau1977_mute() since the core should make sure that this never happens --- include/linux/platform_data/adau1977.h | 45 ++ sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 6 + 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 ++ 7 files changed, 1256 insertions(+) create mode 100644 include/linux/platform_data/adau1977.h 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 diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h new file mode 100644 index 0000000..bed11d9 --- /dev/null +++ b/include/linux/platform_data/adau1977.h @@ -0,0 +1,45 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__ +#define __LINUX_PLATFORM_DATA_ADAU1977_H__ + +/** + * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting + * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V + * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V + * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V + * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V + * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V + * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V + * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V + * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V + * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V + */ +enum adau1977_micbias { + ADAU1977_MICBIAS_5V0 = 0x0, + ADAU1977_MICBIAS_5V5 = 0x1, + ADAU1977_MICBIAS_6V0 = 0x2, + ADAU1977_MICBIAS_6V5 = 0x3, + ADAU1977_MICBIAS_7V0 = 0x4, + ADAU1977_MICBIAS_7V5 = 0x5, + ADAU1977_MICBIAS_8V0 = 0x6, + ADAU1977_MICBIAS_8V5 = 0x7, + ADAU1977_MICBIAS_9V0 = 0x8, +}; + +/** + * struct adau1977_platform_data - Platform configuration data for the ADAU1977 + * @micbias: Specifies the voltage for the MICBIAS pin + */ +struct adau1977_platform_data { + enum adau1977_micbias micbias; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fee1f64..9a5f85e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -24,6 +24,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV801 if SPI_MASTER select SND_SOC_ADAV803 if I2C + select SND_SOC_ADAU1977_SPI if SPI_MASTER + select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -202,6 +204,19 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU1977 + tristate + +config SND_SOC_ADAU1977_SPI + tristate + select SND_SOC_ADAU1977 + select REGMAP_SPI + +config SND_SOC_ADAU1977_I2C + tristate + select SND_SOC_ADAU1977 + select REGMAP_I2C + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index eeb2f91..12dc02c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,9 @@ 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-adau1977-objs := adau1977.o +snd-soc-adau1977-spi-objs := adau1977-spi.o +snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -142,6 +145,9 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(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-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o +obj-$(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 diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c new file mode 100644 index 0000000..9700e8c --- /dev/null +++ b/sound/soc/codecs/adau1977-i2c.c @@ -0,0 +1,59 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static int adau1977_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return adau1977_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + id->driver_data, NULL); +} + +static int adau1977_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id adau1977_i2c_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); + +static struct i2c_driver adau1977_i2c_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_i2c_probe, + .remove = adau1977_i2c_remove, + .id_table = adau1977_i2c_ids, +}; +module_i2c_driver(adau1977_i2c_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c new file mode 100644 index 0000000..b05cf5d --- /dev/null +++ b/sound/soc/codecs/adau1977-spi.c @@ -0,0 +1,76 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static void adau1977_spi_switch_mode(struct device *dev) +{ + struct spi_device *spi = to_spi_device(dev); + + /* + * To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. + */ + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); +} + +static int adau1977_spi_probe(struct spi_device *spi) +{ + const struct spi_device_id *id = spi_get_device_id(spi); + struct regmap_config config; + + if (!id) + return -EINVAL; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x1; + + return adau1977_probe(&spi->dev, + devm_regmap_init_spi(spi, &config), + id->driver_data, adau1977_spi_switch_mode); +} + +static int adau1977_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id adau1977_spi_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); + +static struct spi_driver adau1977_spi_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_spi_probe, + .remove = adau1977_spi_remove, + .id_table = adau1977_spi_ids, +}; +module_spi_driver(adau1977_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c new file mode 100644 index 0000000..fd55da7 --- /dev/null +++ b/sound/soc/codecs/adau1977.c @@ -0,0 +1,1018 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_data/adau1977.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "adau1977.h" + +#define ADAU1977_REG_POWER 0x00 +#define ADAU1977_REG_PLL 0x01 +#define ADAU1977_REG_BOOST 0x02 +#define ADAU1977_REG_MICBIAS 0x03 +#define ADAU1977_REG_BLOCK_POWER_SAI 0x04 +#define ADAU1977_REG_SAI_CTRL0 0x05 +#define ADAU1977_REG_SAI_CTRL1 0x06 +#define ADAU1977_REG_CMAP12 0x07 +#define ADAU1977_REG_CMAP34 0x08 +#define ADAU1977_REG_SAI_OVERTEMP 0x09 +#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x)) +#define ADAU1977_REG_MISC_CONTROL 0x0e +#define ADAU1977_REG_DIAG_CONTROL 0x10 +#define ADAU1977_REG_STATUS(x) (0x11 + (x)) +#define ADAU1977_REG_DIAG_IRQ1 0x15 +#define ADAU1977_REG_DIAG_IRQ2 0x16 +#define ADAU1977_REG_ADJUST1 0x17 +#define ADAU1977_REG_ADJUST2 0x18 +#define ADAU1977_REG_ADC_CLIP 0x19 +#define ADAU1977_REG_DC_HPF_CAL 0x1a + +#define ADAU1977_POWER_RESET BIT(7) +#define ADAU1977_POWER_PWUP BIT(0) + +#define ADAU1977_PLL_CLK_S BIT(4) +#define ADAU1977_PLL_MCS_MASK 0x7 + +#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0 +#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4 + +#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7) +#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6) +#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5) + +#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6) +#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6) +#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6) + +#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3) +#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3) + +#define ADAU1977_SAI_CTRL0_FS_MASK (0x7) +#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0) +#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1) +#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2) +#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3) +#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4) + +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4) +#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3) +#define ADAU1977_SAI_CTRL1_MSB BIT(2) +#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1) +#define ADAU1977_SAI_CTRL1_MASTER BIT(0) + +#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x)) +#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3) + +#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6) +#define ADAU1977_MISC_CONTROL_MMUTE BIT(4) +#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0) + +#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4 +#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0 + +struct adau1977 { + struct regmap *regmap; + bool right_j; + unsigned int sysclk; + enum adau1977_sysclk_src sysclk_src; + struct gpio_desc *reset_gpio; + enum adau1977_type type; + + struct regulator *avdd_reg; + struct regulator *dvdd_reg; + + struct snd_pcm_hw_constraint_list constraints; + + struct device *dev; + void (*switch_mode)(struct device *dev); + + unsigned int max_master_fs; + unsigned int slot_width; + bool enabled; + bool master; +}; + +static const struct reg_default adau1977_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x41 }, + { 0x02, 0x4a }, + { 0x03, 0x7d }, + { 0x04, 0x3d }, + { 0x05, 0x02 }, + { 0x06, 0x00 }, + { 0x07, 0x10 }, + { 0x08, 0x32 }, + { 0x09, 0xf0 }, + { 0x0a, 0xa0 }, + { 0x0b, 0xa0 }, + { 0x0c, 0xa0 }, + { 0x0d, 0xa0 }, + { 0x0e, 0x02 }, + { 0x10, 0x0f }, + { 0x15, 0x20 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x1a, 0x00 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000); + +static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS, + 3, 0, NULL, 0) +}; + +static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI, + 4, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0), + SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0), + SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0), + SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0), + + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + + SND_SOC_DAPM_OUTPUT("VREF"), +}; + +static const struct snd_soc_dapm_route adau1977_dapm_routes[] = { + { "ADC1", NULL, "AIN1" }, + { "ADC2", NULL, "AIN2" }, + { "ADC3", NULL, "AIN3" }, + { "ADC4", NULL, "AIN4" }, + + { "ADC1", NULL, "Vref" }, + { "ADC2", NULL, "Vref" }, + { "ADC3", NULL, "Vref" }, + { "ADC4", NULL, "Vref" }, + + { "VREF", NULL, "Vref" }, +}; + +#define ADAU1977_VOLUME(x) \ + SOC_SINGLE_TLV("ADC" #x " Capture Volume", \ + ADAU1977_REG_POST_ADC_GAIN((x) - 1), \ + 0, 255, 1, adau1977_adc_gain) + +#define ADAU1977_HPF_SWITCH(x) \ + SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0) + +#define ADAU1977_DC_SUB_SWITCH(x) \ + SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0) + +static const struct snd_kcontrol_new adau1977_snd_controls[] = { + ADAU1977_VOLUME(1), + ADAU1977_VOLUME(2), + ADAU1977_VOLUME(3), + ADAU1977_VOLUME(4), + + ADAU1977_HPF_SWITCH(1), + ADAU1977_HPF_SWITCH(2), + ADAU1977_HPF_SWITCH(3), + ADAU1977_HPF_SWITCH(4), + + ADAU1977_DC_SUB_SWITCH(1), + ADAU1977_DC_SUB_SWITCH(2), + ADAU1977_DC_SUB_SWITCH(3), + ADAU1977_DC_SUB_SWITCH(4), +}; + +static int adau1977_reset(struct adau1977 *adau1977) +{ + int ret; + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_RESET); + regcache_cache_bypass(adau1977->regmap, false); + if (ret) + return ret; + + return ret; +} + +/* + * Returns the appropriate setting for ths FS field in the CTRL0 register + * depending on the rate. + */ +static int adau1977_lookup_fs(unsigned int rate) +{ + if (rate >= 8000 && rate <= 12000) + return ADAU1977_SAI_CTRL0_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + return ADAU1977_SAI_CTRL0_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + return ADAU1977_SAI_CTRL0_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + return ADAU1977_SAI_CTRL0_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + return ADAU1977_SAI_CTRL0_FS_128000_192000; + else + return -EINVAL; +} + +static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate, + unsigned int fs) +{ + unsigned int mcs; + + /* + * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs + * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs + * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate) + */ + + rate *= 512 >> fs; + + if (adau1977->sysclk % rate != 0) + return -EINVAL; + + mcs = adau1977->sysclk / rate; + + /* The factors configured by MCS are 1, 2, 3, 4, 6 */ + if (mcs < 1 || mcs > 6 || mcs == 5) + return -EINVAL; + + mcs = mcs - 1; + if (mcs == 5) + mcs = 4; + + return mcs; +} + +static int adau1977_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 adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int slot_width; + unsigned int ctrl0, ctrl0_mask; + unsigned int ctrl1; + int mcs, fs; + int ret; + + fs = adau1977_lookup_fs(rate); + if (fs < 0) + return fs; + + if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) { + mcs = adau1977_lookup_mcs(adau1977, rate, fs); + if (mcs < 0) + return mcs; + } else { + mcs = 0; + } + + ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK; + ctrl0 = fs; + + if (adau1977->right_j) { + switch (params_width(params)) { + case 16: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT; + break; + case 24: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK; + } + + if (adau1977->master) { + switch (params_width(params)) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT; + slot_width = 16; + break; + case 24: + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT; + slot_width = 32; + break; + default: + return -EINVAL; + } + + /* In TDM mode there is a fixed slot width */ + if (adau1977->slot_width) + slot_width = adau1977->slot_width; + + if (slot_width == 16) + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16; + else + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32; + + ret = regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK | + ADAU1977_SAI_CTRL1_BCLKRATE_MASK, + ctrl1); + if (ret < 0) + return ret; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ctrl0_mask, ctrl0); + if (ret < 0) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_MCS_MASK, mcs); +} + +static int adau1977_power_disable(struct adau1977 *adau1977) +{ + int ret = 0; + + if (!adau1977->enabled) + return 0; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, 0); + if (ret) + return ret; + + regcache_mark_dirty(adau1977->regmap); + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 0); + + regcache_cache_only(adau1977->regmap, true); + + regulator_disable(adau1977->avdd_reg); + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); + + adau1977->enabled = false; + + return 0; +} + +static int adau1977_power_enable(struct adau1977 *adau1977) +{ + unsigned int val; + int ret = 0; + + if (adau1977->enabled) + return 0; + + ret = regulator_enable(adau1977->avdd_reg); + if (ret) + return ret; + + if (adau1977->dvdd_reg) { + ret = regulator_enable(adau1977->dvdd_reg); + if (ret) + goto err_disable_avdd; + } + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 1); + + regcache_cache_only(adau1977->regmap, false); + + if (adau1977->switch_mode) + adau1977->switch_mode(adau1977->dev); + + ret = adau1977_reset(adau1977); + if (ret) + goto err_disable_dvdd; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP); + if (ret) + goto err_disable_dvdd; + + ret = regcache_sync(adau1977->regmap); + if (ret) + goto err_disable_dvdd; + + /* + * The PLL register is not affected by the software reset. It is + * possible that the value of the register was changed to the + * default value while we were in cache only mode. In this case + * regcache_sync will skip over it and we have to manually sync + * it. + */ + ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val); + if (ret) + goto err_disable_dvdd; + + if (val == 0x41) { + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL, + 0x41); + if (ret) + goto err_disable_dvdd; + regcache_cache_bypass(adau1977->regmap, false); + } + + adau1977->enabled = true; + + return ret; + +err_disable_dvdd: + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); +err_disable_avdd: + regulator_disable(adau1977->avdd_reg); + return ret; +} + +static int adau1977_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = adau1977_power_enable(adau1977); + break; + case SND_SOC_BIAS_OFF: + ret = adau1977_power_disable(adau1977); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1, drv; + unsigned int slot[4]; + unsigned int i; + int ret; + + if (slots == 0) { + /* 0 = No fixed slot width */ + adau1977->slot_width = 0; + adau1977->max_master_fs = 192000; + return regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK, + ADAU1977_SAI_CTRL0_SAI_I2S); + } + + if (rx_mask == 0 || tx_mask != 0) + return -EINVAL; + + drv = 0; + for (i = 0; i < 4; i++) { + slot[i] = __ffs(rx_mask); + drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i); + rx_mask &= ~(1 << slot[i]); + if (slot[i] >= slots) + return -EINVAL; + if (rx_mask == 0) + break; + } + + if (rx_mask != 0) + return -EINVAL; + + switch (width) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16; + break; + case 24: + /* We can only generate 16 bit or 32 bit wide slots */ + if (adau1977->master) + return -EINVAL; + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24; + break; + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 2: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2; + break; + case 4: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4; + break; + case 8: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8; + break; + case 16: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_C(0) | + ADAU1977_SAI_OVERTEMP_DRV_C(1) | + ADAU1977_SAI_OVERTEMP_DRV_C(2) | + ADAU1977_SAI_OVERTEMP_DRV_C(3), drv); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12, + (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34, + (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1); + if (ret) + return ret; + + adau1977->slot_width = width; + + /* In master mode the maximum bitclock is 24.576 MHz */ + adau1977->max_master_fs = min(192000, 24576000 / width / slots); + + return 0; +} + +static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = ADAU1977_MISC_CONTROL_MMUTE; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL, + ADAU1977_MISC_CONTROL_MMUTE, val); +} + +static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0; + bool invert_lrclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + adau1977->master = false; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ctrl1 |= ADAU1977_SAI_CTRL1_MASTER; + adau1977->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = true; + break; + default: + return -EINVAL; + } + + adau1977->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + adau1977->right_j = true; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = false; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) + block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + ADAU1977_BLOCK_POWER_SAI_LR_POL | + ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_FMT_MASK, + ctrl0); + if (ret) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE, + ctrl1); +} + +static int adau1977_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + u64 formats = 0; + + if (adau1977->slot_width == 16) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE; + else if (adau1977->right_j || adau1977->slot_width == 24) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints); + + if (adau1977->master) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs); + + if (formats != 0) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); + + return 0; +} + +static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (tristate) + val = ADAU1977_SAI_OVERTEMP_DRV_HIZ; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_HIZ, val); +} + +static const struct snd_soc_dai_ops adau1977_dai_ops = { + .startup = adau1977_startup, + .hw_params = adau1977_hw_params, + .mute_stream = adau1977_mute, + .set_fmt = adau1977_set_dai_fmt, + .set_tdm_slot = adau1977_set_tdm_slot, + .set_tristate = adau1977_set_tristate, +}; + +static struct snd_soc_dai_driver adau1977_dai = { + .name = "adau1977-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, + .ops = &adau1977_dai_ops, +}; + +static const unsigned int adau1977_rates[] = { + 8000, 16000, 32000, 64000, 128000, + 11025, 22050, 44100, 88200, 172400, + 12000, 24000, 48000, 96000, 192000, +}; + +#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f +#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0 +#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00 +/* All rates >= 32000 */ +#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c + +static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq) +{ + unsigned int mcs; + + if (mclk % (base_freq * 128) != 0) + return false; + + mcs = mclk / (128 * base_freq); + if (mcs < 1 || mcs > 6 || mcs == 5) + return false; + + return true; +} + +static int adau1977_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = 0; + unsigned int clk_src; + unsigned int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + if (clk_id != ADAU1977_SYSCLK) + return -EINVAL; + + switch (source) { + case ADAU1977_SYSCLK_SRC_MCLK: + clk_src = 0; + break; + case ADAU1977_SYSCLK_SRC_LRCLK: + clk_src = ADAU1977_PLL_CLK_S; + break; + default: + return -EINVAL; + } + + if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) { + if (freq < 4000000 || freq > 36864000) + return -EINVAL; + + if (adau1977_check_sysclk(freq, 32000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000; + if (adau1977_check_sysclk(freq, 44100)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100; + if (adau1977_check_sysclk(freq, 48000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000; + + if (mask == 0) + return -EINVAL; + } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) { + mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_CLK_S, clk_src); + if (ret) + return ret; + + adau1977->constraints.mask = mask; + adau1977->sysclk_src = source; + adau1977->sysclk = freq; + + return 0; +} + +static int adau1977_codec_probe(struct snd_soc_codec *codec) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (adau1977->type) { + case ADAU1977: + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau1977_micbias_dapm_widgets, + ARRAY_SIZE(adau1977_micbias_dapm_widgets)); + if (ret < 0) + return ret; + break; + default: + break; + } + + return 0; +} + +static struct snd_soc_codec_driver adau1977_codec_driver = { + .probe = adau1977_codec_probe, + .set_bias_level = adau1977_set_bias_level, + .set_sysclk = adau1977_set_sysclk, + .idle_bias_off = true, + + .controls = adau1977_snd_controls, + .num_controls = ARRAY_SIZE(adau1977_snd_controls), + .dapm_widgets = adau1977_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets), + .dapm_routes = adau1977_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes), +}; + +static int adau1977_setup_micbias(struct adau1977 *adau1977) +{ + struct adau1977_platform_data *pdata = adau1977->dev->platform_data; + unsigned int micbias; + + if (pdata) { + micbias = pdata->micbias; + if (micbias > ADAU1977_MICBIAS_9V0) + return -EINVAL; + + } else { + micbias = ADAU1977_MICBIAS_8V5; + } + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS, + ADAU1977_MICBIAS_MB_VOLTS_MASK, + micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET); +} + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)) +{ + unsigned int power_off_mask; + struct adau1977 *adau1977; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL); + if (adau1977 == NULL) + return -ENOMEM; + + adau1977->dev = dev; + adau1977->type = type; + adau1977->regmap = regmap; + adau1977->switch_mode = switch_mode; + adau1977->max_master_fs = 192000; + + adau1977->constraints.list = adau1977_rates; + adau1977->constraints.count = ARRAY_SIZE(adau1977_rates); + + adau1977->avdd_reg = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(adau1977->avdd_reg)) + return PTR_ERR(adau1977->avdd_reg); + + adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD"); + if (IS_ERR(adau1977->dvdd_reg)) { + if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV) + return PTR_ERR(adau1977->dvdd_reg); + adau1977->dvdd_reg = NULL; + } + + adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); + if (IS_ERR(adau1977->reset_gpio)) { + ret = PTR_ERR(adau1977->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return PTR_ERR(adau1977->reset_gpio); + adau1977->reset_gpio = NULL; + } + + dev_set_drvdata(dev, adau1977); + + if (adau1977->reset_gpio) { + ret = gpiod_direction_output(adau1977->reset_gpio, 0); + if (ret) + return ret; + ndelay(100); + } + + ret = adau1977_power_enable(adau1977); + if (ret) + return ret; + + if (type == ADAU1977) { + ret = adau1977_setup_micbias(adau1977); + if (ret) + goto err_poweroff; + } + + if (adau1977->dvdd_reg) + power_off_mask = ~0; + else + power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + power_off_mask, 0x00); + if (ret) + goto err_poweroff; + + ret = adau1977_power_disable(adau1977); + if (ret) + return ret; + + return snd_soc_register_codec(dev, &adau1977_codec_driver, + &adau1977_dai, 1); + +err_poweroff: + adau1977_power_disable(adau1977); + return ret; + +} +EXPORT_SYMBOL_GPL(adau1977_probe); + +static bool adau1977_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1977_REG_STATUS(0): + case ADAU1977_REG_STATUS(1): + case ADAU1977_REG_STATUS(2): + case ADAU1977_REG_STATUS(3): + case ADAU1977_REG_ADC_CLIP: + return true; + } + + return false; +} + +const struct regmap_config adau1977_regmap_config = { + .max_register = ADAU1977_REG_DC_HPF_CAL, + .volatile_reg = adau1977_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1977_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults), +}; +EXPORT_SYMBOL_GPL(adau1977_regmap_config); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h new file mode 100644 index 0000000..95e7143 --- /dev/null +++ b/sound/soc/codecs/adau1977.h @@ -0,0 +1,37 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SOUND_SOC_CODECS_ADAU1977_H__ +#define __SOUND_SOC_CODECS_ADAU1977_H__ + +#include <linux/regmap.h> + +struct device; + +enum adau1977_type { + ADAU1977, + ADAU1978, + ADAU1979, +}; + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)); + +extern const struct regmap_config adau1977_regmap_config; + +enum adau1977_clk_id { + ADAU1977_SYSCLK, +}; + +enum adau1977_sysclk_src { + ADAU1977_SYSCLK_SRC_MCLK, + ADAU1977_SYSCLK_SRC_LRCLK, +}; + +#endif -- 1.8.0

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[alsa-devel] [PATCH] ASoC: dapm: Correct regulator bypass error messages
by Charles Keepax 19 Feb '14

19 Feb '14

The error messages for bypassing/unbypassing a regulator appear to be swapped round, this patch corrects these. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c2d751..7e10be8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1219,7 +1219,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to bypass %s: %d\n", + "ASoC: Failed to unbypass %s: %d\n", w->name, ret); } @@ -1229,7 +1229,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3249,7 +3249,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; -- 1.7.2.5

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[alsa-devel] [PATCHv2] ASoC: Intel: Rename SST trace event header to be less generic.
by Liam Girdwood 19 Feb '14

19 Feb '14

The Intel audio DSP SST trace event header has been renamed from sst.h to intel-sst.h in order to avoid any confusion with any future Samoa Standard Time drivers ;) Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/trace/events/{sst.h => intel-sst.h} | 6 +++--- sound/soc/intel/sst-dsp.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) rename include/trace/events/{sst.h => intel-sst.h} (94%) diff --git a/include/trace/events/sst.h b/include/trace/events/intel-sst.h similarity index 94% rename from include/trace/events/sst.h rename to include/trace/events/intel-sst.h index 1a0d6b3..76c72d3 100644 --- a/include/trace/events/sst.h +++ b/include/trace/events/intel-sst.h @@ -1,8 +1,8 @@ #undef TRACE_SYSTEM -#define TRACE_SYSTEM sst +#define TRACE_SYSTEM intel-sst -#if !defined(_TRACE_SST_H) || defined(TRACE_HEADER_MULTI_READ) -#define _TRACE_SST_H +#if !defined(_TRACE_INTEL_SST_H) || defined(TRACE_HEADER_MULTI_READ) +#define _TRACE_INTEL_SST_H #include <linux/types.h> #include <linux/ktime.h> diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 1888de5..e0ad2e5 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -24,7 +24,7 @@ #include "sst-dsp-priv.h" #define CREATE_TRACE_POINTS -#include <trace/events/sst.h> +#include <trace/events/intel-sst.h> /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) -- 1.8.3.2

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[alsa-devel] [PATCH 1/2] ASoC: Intel: Rename SST trace event header to be less generic.
by Liam Girdwood 19 Feb '14

19 Feb '14

The Intel audio DSP SST trace event header has been renamed from sst.h to intel-sst.h in order to avoid any confusion with any future Samoa Standard Time drivers ;) Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/trace/events/{sst.h => intel-sst.h} | 0 sound/soc/intel/sst-dsp.c | 2 +- 2 files changed, 1 insertion(+), 1 deletion(-) rename include/trace/events/{sst.h => intel-sst.h} (100%) diff --git a/include/trace/events/sst.h b/include/trace/events/intel-sst.h similarity index 100% rename from include/trace/events/sst.h rename to include/trace/events/intel-sst.h diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 1888de5..e0ad2e5 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -24,7 +24,7 @@ #include "sst-dsp-priv.h" #define CREATE_TRACE_POINTS -#include <trace/events/sst.h> +#include <trace/events/intel-sst.h> /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) -- 1.8.3.2

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[alsa-devel] [PATCH 1/1] ASoC: cs42l51: Remove unused variable
by Sachin Kamat 19 Feb '14

19 Feb '14

‘cs42l51’ is not used. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs42l51.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e53c8714591f..3eab46020a30 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -487,7 +487,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { - struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); -- 1.7.9.5

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[alsa-devel] Failed RPM build RHEL 6.4 /boot/ vmlinuz-2.6.32-279.22.1.el6.x86_64
by Ionut V. 18 Feb '14

18 Feb '14

Not too much of a Linux user but i got this when trying to install alsa on RHEL 6.4 with kernel - 2.6.32-279.22.1.el6.x86_64 Hope attachments work, if not i will paste the whole output in the replay. -- Ionut V.

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[alsa-devel] SOC_VALUE_ENUM usages in cs42l52 & cs42l73 drivers
by Takashi Iwai 18 Feb '14

18 Feb '14

Hi, as I audited through ASoC codec drivers recently, I noticed that cs42l52 and cs42l73 drivers have some strange SOC_VALUE_ENUM usages. For example, cs42l52.c has the following code snippet: static const char * const left_swap_text[] = { "Left", "LR 2", "Right"}; .... static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); Note that the third argument of SOC_VALUE_ENUM_SINGLE() is the mask bits. Passing 1 means only the least bit is evaluated no matter what value is assigned while swap_values can be 0, 1 or 3. I suppose here meant 3, but would like to know how the code is supposed to work. There are three value-enum definitions in cs42l52.c, and all pass 1 there. Similarly, cs42l73.c has four definitions, but with different masks: SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, ... while values can be either 0, 1 or 2. Something really fishy here. thanks, Takashi

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[alsa-devel] [PATCH 0/2] alsa-lib surround 2.1 support
by David Henningsson 18 Feb '14

18 Feb '14

I'm currently experimenting with different ways to make the system autodetect if surround2.1 is available through a four channel setup (e g laptop surround speaker) and fallback to 5.1 if it isn't. It's a bit tricky, especially given that the LFE speaker could be on left channel, right channel, and both, so meanwhile, this fallback should be better than nothing and help establish a standard for upper parts of the stack (i e PulseAudio). David Henningsson (2): conf: Add basic infrastructure for 2.1 surround sound conf: Add surround 2.1 support to all cards src/conf/alsa.conf | 3 ++ src/conf/cards/ATIIXP-SPDMA.conf | 1 + src/conf/cards/ATIIXP.conf | 1 + src/conf/cards/Audigy.conf | 1 + src/conf/cards/Audigy2.conf | 1 + src/conf/cards/Aureon51.conf | 1 + src/conf/cards/Aureon71.conf | 1 + src/conf/cards/CA0106.conf | 1 + src/conf/cards/CMI8738-MC6.conf | 1 + src/conf/cards/CMI8738-MC8.conf | 1 + src/conf/cards/CMI8788.conf | 2 +- src/conf/cards/CS46xx.conf | 1 + src/conf/cards/EMU10K1.conf | 1 + src/conf/cards/EMU10K1X.conf | 1 + src/conf/cards/FM801.conf | 1 + src/conf/cards/FireWave.conf | 1 + src/conf/cards/HDA-Intel.conf | 1 + src/conf/cards/ICE1712.conf | 1 + src/conf/cards/ICE1724.conf | 1 + src/conf/cards/ICH.conf | 1 + src/conf/cards/ICH4.conf | 1 + src/conf/cards/Loopback.conf | 1 + src/conf/cards/NFORCE.conf | 1 + src/conf/cards/SB-XFi.conf | 1 + src/conf/cards/SI7018.conf | 1 + src/conf/cards/USB-Audio.conf | 1 + src/conf/cards/VIA8233.conf | 1 + src/conf/cards/VIA8233A.conf | 1 + src/conf/cards/VIA8237.conf | 1 + src/conf/pcm/Makefile.am | 2 +- src/conf/pcm/surround21.conf | 62 ++++++++++++++++++++++++++++++++++++++++ 31 files changed, 94 insertions(+), 2 deletions(-) create mode 100644 src/conf/pcm/surround21.conf -- 1.9.rc1

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