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[alsa-devel] [PATCH 0/2] OMAPDSS: Couple of HDMI audio fixes
by Jyri Sarha 28 Aug '15

28 Aug '15

The first fix is a hairy one, but I think the locking is fool proof now. It is needed because there is no telling in which order user space starts an audio and video stream playback. If the audio is started first and the video mode is changed when video playback starts the audio setup needs to survive display turning off and back on again. After this patch the audio streams should survive a suspend-resume cycle too. The second one is a trivial work-around to a problem in ALSA constraint resolver code. The two fixes are totally independent and the video and audio side patches applied separately trough their own trees. Jyri Sarha (2): OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled ASoC: omap-hdmi-audio: Set buffer bytes step constraint to 128 drivers/video/fbdev/omap2/dss/hdmi.h | 10 +++- drivers/video/fbdev/omap2/dss/hdmi4.c | 65 ++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 89 +++++++++++++++++++++++++++++------ sound/soc/omap/omap-hdmi-audio.c | 10 +++- 4 files changed, 146 insertions(+), 28 deletions(-) -- 1.9.1

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[alsa-devel] Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 28 Aug '15

28 Aug '15

Hi Mika, Dan et al. @Dan : thank you, I will use get_maintainer.pl as of now. @Mika: you mean I should prepare a new DSDT table and in this case overwrite the SPI value there with the 0 and load it (the DSDT table ) into a custom kernel build ?? I will do this (cause all what I want is to enable the codec and to hear sound) currently I have in the extracted DSDT table from this device the SPI1 controller for the "AUDI" device as seen in this snippet : Scope (_SB) { Device (LPEA) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "17AA7004") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_DEP, Package (0x01) // _DEP: Dependencies { ^SPI1.AUDI }) Name (_PR0, Package (0x01) // _PR0: Power Resources for D0 { PLPE }) .... some lines later follows the AUDI device with Device (AUDI) { Name (_HID, "WM510205") // _HID: Hardware ID Name (_CID, "WM510205") // _CID: Compatible ID Name (_DDN, "Wolfson Microelectronics Audio WM5102") // _DDN: DOS Device Name Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveLow, ExclusiveAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.I2C7.PMIC", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO1", 0x00, ResourceConsumer, , ) { // Pin list 0x0017 } }) Return (SBUF) /* \_SB_.SPI1.AUDI._CRS.SBUF */ } .... I t makes sense to me, to change the current value(s) in the DSDT and I have read about it to use a custom DSDT table but I am not 100% sure yet (until I have tested it). If I understand you correct, than all I have to do is to change the 'SPI1' string in this dsdt to SPI0 like ^SPI1.AUDI to ^SPI0.AUDI and this block here SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , becomes SpiSerialBus (0x0000, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI0", 0x00, ResourceConsumer, , For this case, could you please have a look in the whole DSDT table that I extract from this device? If nothing else seems relevant and the current system dsdt has to be changed as suggested, I will try this out as soon as possible @Jarkko : as you suggested , I attach the current system DSDT as a binary. I have also disassembled, and if someone needs it, I can also paste it somewhere else as a reference After all I will assembly the new dsdt and include it in my custom kernel builld as documented, but will wait for some new infos etc. thank you for you answers @all cheers chris 2015-06-25 9:36 GMT+02:00 Mika Westerberg <mika.westerberg(a)linux.intel.com>: > On Thu, Jun 25, 2015 at 09:29:31AM +0200, Christian Hartmann wrote: >> Hello Maintainer(s)s of SPI, >> >> hope I got all of you maintainers now. This email was already sent out >> last week to the alsa-devel list and I resent it here to you. >> >> I have one patch that enables the spi initialization of the WM510205 audio codec >> Without the patch, this message indicates that the codec chip fails >> with the current chipselect number in spi-pxa2xx: >> >> >> [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 >> [ 0.296270] spi_master spi32766: failed to add SPI device >> WM510205:00 from ACPI >> >> So raise num_chipselect by one to enable the codec. The WM510202 is on >> chipselect = 1. > > The hardware only has one chipselect. > > You can fix this by specifying the correct chipselect in the ACPI > SpiSerialBus connector: > > SpiSerialBus (0, PolarityLow, FourWireMode, 8, ControllerInitiated, .. > > First parameter is the chipselect.

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[alsa-devel] [PATCH] ALSA: usb-audio: correct the value cache check.
by Yao-Wen Mao 28 Aug '15

28 Aug '15

The check of cval->cached should be zero-based (including master channel). Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd5c49e..f494dce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2540,7 +2540,7 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) for (c = 0; c < MAX_CHANNELS; c++) { if (!(cval->cmask & (1 << c))) continue; - if (cval->cached & (1 << c)) { + if (cval->cached & (1 << (c + 1))) { err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) -- 2.5.0.457.gab17608

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[alsa-devel] [PATCH v2] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning
by Oder Chiou 28 Aug '15

28 Aug '15

Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning. It de-initials the jack detection function, and it cannot be turned off in _BIAS_OFF. If we don't de-initial it, the pop sound will be heard in the situation of powering off. And replace the related register settings from magic number to meaningful defined name. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 25 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 50f797c..4972bf3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2806,13 +2806,13 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0004); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); msleep(100); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); @@ -2836,10 +2836,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } else { /* jack out */ rt5645->jack_type = 0; - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, - 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, 0); if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); @@ -3458,13 +3458,26 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_RING2_SLEEVE_GND, RT5645_RING2_SLEEVE_GND); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, + RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, + 0); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index dffdda2..0e4cfc6 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2115,6 +2115,7 @@ enum { #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) +#define RT5645_RING2_SLEEVE_GND (0x1 << 5) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Applied "ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers" to the asoc tree
by Mark Brown 27 Aug '15

27 Aug '15

The patch ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 054bc835d27b558393541c32a209c01d89cda75a Mon Sep 17 00:00:00 2001 From: Luis de Bethencourt <luis(a)debethencourt.com> Date: Thu, 27 Aug 2015 17:04:01 +0200 Subject: [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/bcm2835-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cbf8ec1..8c435beb263d 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { -- 2.5.0

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[alsa-devel] [PATCH v5] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 27 Aug '15

27 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Tested on a setup with a single ICS-43432 (the device itself supports stereo operation using a hardware pin controlling left vs. right channel output). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V5: Not a lot left on the code now... Removed set_dai_fmt as it only checked the formats and never did any useful work. V4: Updated commit message that the driver has been tested with real hardware, after off-list comment on V3. V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 +++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 77 ++++++++++++++++++++ 5 files changed, 101 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..4f202c1 --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,77 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] Applied "ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning" to the asoc tree
by Mark Brown 27 Aug '15

27 Aug '15

The patch ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f2988afedf2c19880a3c65d79dfc7939e1b53d8a Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Thu, 27 Aug 2015 15:14:51 +0800 Subject: [PATCH] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning. It de-initials the jack detection function, and it cannot be turned off in _BIAS_OFF. If we don't de-initial it, the pop sound will be heard in the situation of powering off. And replace the related register settings from magic number to meaningful defined name. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 25 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 0a4cb6bc3433..db50b035b00b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2806,13 +2806,13 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0004); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); msleep(100); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); @@ -2836,10 +2836,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } else { /* jack out */ rt5645->jack_type = 0; - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, - 0x1000); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, - 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, 0); if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); @@ -3449,6 +3449,18 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_RING2_SLEEVE_GND, RT5645_RING2_SLEEVE_GND); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, + RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, + 0); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", @@ -3456,7 +3468,8 @@ static struct i2c_driver rt5645_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 199b22fe1bf3..1987eb825e41 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2111,6 +2111,7 @@ enum { #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) +#define RT5645_RING2_SLEEVE_GND (0x1 << 5) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 2.5.0

1 0

[alsa-devel] [PATCH 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 27 Aug '15

27 Aug '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 +++++++++++++ 2 files changed, 255 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..71b84cd 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,184 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2894,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

2 3

[alsa-devel] [PATCH v2 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 27 Aug '15

27 Aug '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 +++++++++++++ 2 files changed, 255 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..71b84cd 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,184 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2894,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

1 1

[alsa-devel] Out-of-tree HDMI audio driver for BayTrail-T
by David Henningsson 27 Aug '15

27 Aug '15

I don't usually deal with out-of-tree drivers, but when Canonical enabled the Intel Compute Stick [1] (which is using a BayTrail-T processor), we were using an out-of-tree driver to enable HDMI audio. The driver has earlier been made public by Intel [2], and I've put my modifications to it here [3]. Unfortunately, it looks like the effort and hardware knowledge required to make this driver upstreamable is beyond what I can currently manage. Also, I cannot answer for whether or not other parties (such as Intel) plan to upstream the driver. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://www.intel.com/content/www/us/en/compute-stick/intel-compute-stick.ht… [2] https://github.com/01org/baytrailaudio [3] http://kernel.ubuntu.com/git/diwic/baytrail-hdmi-audio.git/log/?h=baytrail-…

2 1

[alsa-devel] [PATCH v5 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 27 Aug '15

27 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

5 18

[alsa-devel] Asus X553M audio issue SOLVED
by Rafael Gonzalez 27 Aug '15

27 Aug '15

Guys, I was having a nice chat with debianuser from #alsa on freenode, and he helped me fix my audio issues with my laptop that dual boots windows and linux. So, the easiest way to show you is post the log file of our conversation. ud> Hi there. You've been helping me with my card issues. <salviadud> I want to share what progress I've got. <salviadud> The machine is a dual boot laptop. <salviadud> I also had another issue with ACPI, but I solved that by turning D-Bus on <salviadud> In fact, after d-bus, i can see the d-bus message when I plug in my headphones to the jack. <salviadud> I don't think that windows has been tampering with the soundcard so that I need to do a full electrial reset on it. <salviadud> I just think the alsa driver is ok, but there's something missing. <salviadud> I need to develop it, or something. <salviadud> I want to know who should I talk to. <salviadud> The card vendor (Realtek) <salviadud> The laptop manufacturer (asus) <salviadud> ? <salviadud> If you got any ideas, I am all ears. so to speak <salviadud> And my logic is this: there's no default option to choose between headphones/headsets/speakers. And like I said before, I can only get audio on headphones with 3 ringed jacks. <salviadud> Windows on the other hand, has this nifty userspace program from realtek... I don't like crabs today. <debianuser> Have you managed to boot any other linux liveusb on that laptop? Ubuntu? Fedora? Mageia? Debian? <salviadud> Funny that you mention that <salviadud> I did try to load a kubuntu 14.10 <salviadud> but it hanged on me <salviadud> Same thing happened with mint. <salviadud> So, the other live cd I could load was a slax live usb stick. <salviadud> But that one in particular loaded a different driver <salviadud> and the sound that came out, was crackled. <salviadud> I can still get sound on this thing, but it has to be from external speakers, and hdmi out <salviadud> Although, I think I need to play with hda analyzer for that. <debianuser> I thought that internal speakers work fine, and you only can't hear the sound from headphones... <salviadud> Sorry <salviadud> I meant what you just said <salviadud> I called them external by mistake <salviadud> laptop speakers work, headphones don't <debianuser> Ah, ok :) <salviadud> So, I want to be proactive about this <salviadud> In what other way can I debug this <salviadud> so that I can finally reverse engineer it <salviadud> I think its an unusual thing to have a jack decide on what to output based on the number of rings. <salviadud> I guess I'm old. <salviadud> I'm used to left-right jacks. <salviadud> They are simple pieces of hardware <salviadud> And well, I just love linux, I can't stand windows as my main. <salviadud> Especially for music. <debianuser> It's just there doesn't seem to be anything special about your laptop or codec. It looks normal, but it behaves unusual, as when your mixer values were different from codec dump... <debianuser> Have you tried "full poweroff" by the way? <salviadud> I haven't done it, I need to take out some screws for that. <debianuser> What for? <salviadud> The battery <salviadud> Its not plug-n-play <salviadud> its inside where the hdd lies <debianuser> Ah, well, try just shut it down and unplug power supply, without removing the battery <debianuser> It's just has to be poweroff, not suspend, not hibernate... <salviadud> I'm pretty sure I've done that. <salviadud> start it up on just the battery. <debianuser> (actually sometimes suspend or hibernate may help too, for a different reason, but it's better to try poweroff first...) <salviadud> I'll do it when I get off work <salviadud> Still, getting ahead of it. Lets say it stays the same. <debianuser> The important part is not how you start it, but to power it off for a few minutes, then boot linux first. You can boot it with power supply attached, just boot linux first :) <salviadud> mmmmm <salviadud> Maybe its that <salviadud> the missing link <salviadud> I have never done that <salviadud> after a certain time <salviadud> windows doesn't allow me to access uefi <salviadud> and it boots directly into windows <salviadud> but <salviadud> its because its the first option <salviadud> I just need to change that <salviadud> Ok, let me rephrase <salviadud> If I shut down on linux <salviadud> and I dunno, 10 minutes pass <salviadud> I turn on my computer again, and I want to change my boot option by pressing ESC. <salviadud> I can't <salviadud> It goes into fast boot mode <salviadud> I can't control it <salviadud> So, if I change my boot option <salviadud> I should be able to do that. <salviadud> Ok, please excuse my long explanation. <debianuser> (it was reported to help for different laptops, including laptops having the same codec: launchpad.net/bugs/1087428 comment 4) <debianuser> How do you get to the linux then? <debianuser> I mean how can you boot linux at all? <salviadud> The first time was so hard. <salviadud> I had to go into repair mode in windows. <salviadud> So I could get into the bios. <salviadud> After that, I could choose the cdrom option for boot <salviadud> and that's how I installed it. <salviadud> I don't use a boot manager like grub or lilo <salviadud> just the Bios utility <salviadud> so <salviadud> I usually boot into windows first. Then I do restart, and then it allows me to select my different boot options <salviadud> Its so annoying <salviadud> I'm definetely changing the boot order <debianuser> What if you select restart in windows, get into that options list, and then just press the power button? <salviadud> That link you gave me <salviadud> same exact issue <salviadud> Its too much of a hazzle <salviadud> I want a permanent solution <debianuser> What if you select restart in windows, get into that options list, and then just press the power button - I mean does the laptop turns off? <salviadud> Yeah, it should <salviadud> but when I turn it on, its gonna know it was powered off for a while <salviadud> and it will go into the first boot option <salviadud> which is windows <debianuser> Hm... Is there some hotkey to stop it? Maybe you need to hold Ctrl or Alt? <salviadud> Nope, its NSA loving spyware windows <salviadud> what did you expect? hehehe <salviadud> I know, its terrible <debianuser> Is this the fastboot?: https://www.asus.com/support/FAQ/1005535/ <salviadud> My bios does not look as fancy as that one <salviadud> but yeah, that's the option. <salviadud> I think I had to disable it just to get access to other boot options <salviadud> I think that bios might be for a desktop <salviadud> I dunno, this laptop just feels really secure tight about working on windows <salviadud> and not on linux <salviadud> I'll give it a go in a couple of hourse, I'll tell ya what happened tomorrow <salviadud> U are a lot of help dude, thank you much. <debianuser> It's just its default setting. It may be useful for windows, I mean regular users often don't need boot menu, but they want OS to boot faster :) (I remember how people were impressed when I booted simple linux gui in 10 seconds). <salviadud> 10 seconds? that's still impressive, hehe <salviadud> u know what, I'm gonna try this now <salviadud> brb <salviadud> dude, it totally worked <salviadud> I'm listening to my tunes on audacious <debianuser> Great! :) <salviadud> as a token of my appreciation <salviadud> here's my ftp server <salviadud> anachronox.dyndns.za.org <salviadud> user:ftpuser <salviadud> pass:meridian2002 <salviadud> its full of music roms n pr0n <salviadud> hehe <debianuser> As for 10 seconds boot - it's just a generic kernel, with a custom init script: about 5 seconds to probe/load modules for available devices, 2 seconds for Xorg to start and switch to graphical mode, and 3 more seconds to boot firefox (I don't remember exact timings but something like that). <debianuser> There're better boot times on youtube: https://youtu.be/whkmChncjRk https://youtu.be/XcbKe3ct610 https://youtu.be/fMM-ZZijRc4 https://youtu.be/s7NxCM8ryF8 3-5 seconds - that's impressive. :) But I was too lazy to build a custom kernel for that. :) <salviadud> I could do something similar... <salviadud> I got a custom kernel <salviadud> whatever... I don't care about boot times. <salviadud> thanx again man. <debianuser> You're welcome! Feel free to ask whenever you want to know anything else :) <salviadud> that ftp server gets shutdown when I got to sleep, keep that in mind. <salviadud> I turn it on, every now and then. * debianuser saves a not that full poweroff makes sound headphones working on a dual-boot Asus X553MA (Codec: Realtek ALC270) <debianuser> *a note <debianuser> And thanks for reporting the problem, by the way. If you want to spend a few more minutes for it we can try to compare working and broken settings to check what exactly was different... Maybe we find a way to undo that... <debianuser> (basically I'll just ask you to dump alsa-info 2 more times with one special optio) <debianuser> (and if we're lucky we'll have something to report to alsa-devel mailing list, and devs would be able to include that into the kernel to make it working in next kernel version out of box) <salviadud> ok <salviadud> sounds good <salviadud> How do I run the script, headphones on or off? <debianuser> There're vendor-specific coefficients that control the chip but sometimes are not reset on reboot. To include those int alsa-info run: `echo 1 | sudo tee /sys/module/snd_hda_codec/parameters/dump_coef` then dump alsa-info as usual with headphones plugged in. That's the "working" dump. <debianuser> Then we'll break it: reboot into windows, reboot back into linux, make sure headphones don't work, run `echo 1 | sudo tee /sys/module/snd_hda_codec/parameters/dump_coef` and dump alsa-info with headphones plugged in again. That would be "broken" dump. <salviadud> ok, I don't sudo, so I'll just run that as root. <debianuser> Then we'll diff them and see if there's some difference that we can undo... <salviadud> ok, working dump coming <debianuser> yeah `echo 1 > /sys/module/snd_hda_codec/parameters/dump_coef` as root will work too <salviadud> http://www.alsa-project.org/db/?f=abda2cd2c9137fbf2319015707cf2d8d30aaf773 <salviadud> I ran alsa-info after the echo <salviadud> let me go back into windows..., and see if it breaks on reboot * Registro cargado desde Wed Aug 26 18:45:41 2015 <salviadud> ok, I got broken audio <debianuser> Now let's get broken alsa-info and diff them! <salviadud> http://www.alsa-project.org/db/?f=4d597f8a6f9eaa0f0823d195d5fa5093ea71d0eb <salviadud> Well, I'm glad it worked <salviadud> and hell, I even left windows as a special option <salviadud> no need to erase boot options <salviadud> I just disabled it so I can boot straight into linux <salviadud> feels great man. <debianuser> Hah! The difference is just in 2 coefficients: Coeff 0x04: 0x9827 (working) / 0x9027 (broken) and Coeff 0x0d: 0x4440 (working) / 0xc440 (broken) <debianuser> Let's try to change them back and see if that would fix headphones! <salviadud> so, How do I do that? <debianuser> To do that install "alsa-tools" package if you don't have it yet and run these 4 commands in the following order: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9827 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x0d <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x4440 <salviadud> i got alsatools, let me give it a go. <debianuser> (not sure what which one of them is doing, so let's set them both and see if that will change anything) <salviadud> I got sound <salviadud> You are pro <salviadud> although, I really wouldn't like to do that every time. * salviadud knows that sounded dumb <debianuser> Great! You don't have to do that every time (well, you can add those to /etc/rc.local, or whatever is used in gentoo instead). But the point was to find what was causing the issue. <salviadud> that's why we're doing this test... hehe <salviadud> Yeah, I think its a small, yet important contribution. * debianuser wonders which one of those coefs breaks the sound... <salviadud> I think its safe to just keep both like that. <debianuser> Can you check that by the way? For example revert one of them back to "broken" value and check if you still have the sound from headphones: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9027 <salviadud> I will <salviadud> for science <salviadud> that broke it <salviadud> I got no sound <salviadud> What else do I do? <salviadud> or is that it? <debianuser> How about the other way around? 0x04 to working, 0x0d to broken: <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x04 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x9827 <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 0x0d <debianuser> sudo hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0xc440 <debianuser> (that's the last test) <salviadud> it works <salviadud> I think the other coefficient might be the hdmi <salviadud> But I can't debug that one over here right now <salviadud> still, that's just a wild guess <salviadud> Well, you got all you need to report debugging info? <debianuser> Well, that's all we need for the report. Now if you still feel the power to make the world better (just a little bit) - you can report that to alsa-devel mailing list. Include both alsa-info links, tell which one of them works and which one has headphones broken after windows reboot, and which hda-verb commands pair makes headphones working again. :) <salviadud> I'm just gonna send the log of this conversation <debianuser> They may ask you to test a kernel patch (they often do that) before sending it upstream to Linus. That's why it's better for you to report it. <salviadud> if you don't minnd <debianuser> I don't :) <salviadud> alsa rocks! <debianuser> especially when it works :) <salviadud> \O/ <debianuser> Thank you for your patience, for reporting the problem and for doing all those checks!

1 0

[alsa-devel] different acpi_ipc_irq_index among baytrail boards issue
by Michele Curti 26 Aug '15

26 Aug '15

Hi, some baytrail machines, although belonging to the same platform, have the ADSP IRQ number at different ACPI index (.acpi_ipc_irq_index = 0 instead of 5). Not an issue for now, since there is no support for these machines (like the Asus X205TA) at the moment, but if the driver will work, a patch like the following would be acceptable (I ask so I can try to keep a sort of patchset)? * declare a new struct type sst_acpi_info where to place acpi infos lpe_res_index, ddr_index, ipc_irq_index * create a new 'const struct sst_acpi_info *acpi_info' field into the struct sst_platform_info (to be managed like the others ipc_info, res_info, ...) * move the acpi infos from the byt_rvp_res_info variable to a new byt_acpi_res_info one In this way one could duplicate byt_rvp_acpi_info instead of the whole byt_rvp_res_info (which seems "not good" to me, or it's ok?) to manage different acpi indexes. Hope I was clear enough.. Thanks, Michele --- arch/x86/include/asm/platform_sst_audio.h | 11 ++++++++--- sound/soc/intel/atom/sst/sst_acpi.c | 17 +++++++++++------ 2 files changed, 19 insertions(+), 9 deletions(-) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 7249e6d..b10f6a7 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -118,9 +118,6 @@ struct sst_res_info { unsigned int dram_size; unsigned int mbox_offset; unsigned int mbox_size; - unsigned int acpi_lpe_res_index; - unsigned int acpi_ddr_index; - unsigned int acpi_ipc_irq_index; }; struct sst_ipc_info { @@ -128,13 +125,21 @@ struct sst_ipc_info { unsigned int mbox_recv_off; }; +struct sst_acpi_info { + unsigned int lpe_res_index; + unsigned int ddr_index; + unsigned int ipc_irq_index; +}; + struct sst_platform_info { const struct sst_info *probe_data; const struct sst_ipc_info *ipc_info; + const struct sst_acpi_info *acpi_info; const struct sst_res_info *res_info; const struct sst_lib_dnld_info *lib_info; const char *platform; }; + int add_sst_platform_device(void); #endif diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index bb19b58..427e404 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -116,14 +116,18 @@ static const struct sst_res_info byt_rvp_res_info = { .dram_size = 0x28000, .mbox_offset = 0x144000, .mbox_size = 0x1000, - .acpi_lpe_res_index = 0, - .acpi_ddr_index = 2, - .acpi_ipc_irq_index = 5, +}; + +static const struct sst_acpi_info byt_rvp_acpi_info = { + .lpe_res_index = 0, + .ddr_index = 2, + .ipc_irq_index = 5, }; static struct sst_platform_info byt_rvp_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, + .acpi_info = &byt_rvp_acpi_info, .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", @@ -135,6 +139,7 @@ static struct sst_platform_info byt_rvp_platform_data = { static struct sst_platform_info chv_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, + .acpi_info = &byt_rvp_acpi_info, .lib_info = &byt_lib_dnld_info, .res_info = &byt_rvp_res_info, .platform = "sst-mfld-platform", @@ -148,7 +153,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* All ACPI resource request here */ /* Get Shim addr */ rsrc = platform_get_resource(pdev, IORESOURCE_MEM, - ctx->pdata->res_info->acpi_lpe_res_index); + ctx->pdata->acpi_info->lpe_res_index); if (!rsrc) { dev_err(ctx->dev, "Invalid SHIM base from IFWI"); return -EIO; @@ -202,7 +207,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->mailbox_add = ctx->info.mailbox_start; rsrc = platform_get_resource(pdev, IORESOURCE_MEM, - ctx->pdata->res_info->acpi_ddr_index); + ctx->pdata->acpi_info->ddr_index); if (!rsrc) { dev_err(ctx->dev, "Invalid DDR base from IFWI"); return -EIO; @@ -219,7 +224,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, - ctx->pdata->res_info->acpi_ipc_irq_index); + ctx->pdata->acpi_info->ipc_irq_index); return 0; } -- 2.5.0

1 0

[alsa-devel] [PATCH 0/3] USB-audio disconnection lockdep fixes
by Takashi Iwai 26 Aug '15

26 Aug '15

Hi, this is a series of patches for USB-audio to improve the lock handling at disconnection. Originally it was aimed to get rid of spurious lockdep warnings, but this ended up with the rewrite and improvement of locking code at disconnection and its checks in the mixer and PCM codes. Takashi === Takashi Iwai (3): ALSA: usb-audio: Avoid nested autoresume calls ALSA: usb-audio: Replace probing flag with active refcount ALSA: usb-audio: Handle normal and auto-suspend equally sound/usb/card.c | 108 ++++++++++++++++++++++------------------ sound/usb/endpoint.c | 10 ++-- sound/usb/mixer.c | 32 ++++-------- sound/usb/mixer_quirks.c | 126 ++++++++++++++++++++--------------------------- sound/usb/pcm.c | 32 ++++++------ sound/usb/proc.c | 4 +- sound/usb/usbaudio.h | 11 +++-- 7 files changed, 155 insertions(+), 168 deletions(-) -- 2.5.0

1 3

[alsa-devel] ALSA/usb-audio: snd_usb_autoresume possible recursive locking detected
by Alexnader Kuleshov 26 Aug '15

26 Aug '15

Hello all, Today I've update to 4.2.0-rc8+ and started to get following lines in the dmesg output: [ 13.884092] ============================================= [ 13.884092] [ INFO: possible recursive locking detected ] [ 13.884094] 4.2.0-rc8+ #61 Not tainted [ 13.884094] --------------------------------------------- [ 13.884095] pulseaudio/980 is trying to acquire lock: [ 13.884096] (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884103] but task is already holding lock: [ 13.884104] (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884107] other info that might help us debug this: [ 13.884108] Possible unsafe locking scenario: [ 13.884109] CPU0 [ 13.884110] ---- [ 13.884110] lock(&chip->shutdown_rwsem); [ 13.884111] lock(&chip->shutdown_rwsem); [ 13.884112] *** DEADLOCK *** [ 13.884113] May be due to missing lock nesting notation [ 13.884114] 2 locks held by pulseaudio/980: [ 13.884115] #0: (&pcm->open_mutex){+.+.+.}, at: [<ffffffffa02ac9bc>] snd_pcm_open+0xa9/0x1f8 [snd_pcm] [ 13.884120] #1: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884124] stack backtrace: [ 13.884125] CPU: 0 PID: 980 Comm: pulseaudio Not tainted 4.2.0-rc8+ #61 [ 13.884126] Hardware name: Gigabyte Technology Co., Ltd. Z97X-UD5H-BK/Z97X-UD5H-BK, BIOS F6 06/17/2014 [ 13.884127] 0000000000000000 000000003d4c66e6 ffff88040897b4d8 ffffffff815ba622 [ 13.884129] 0000000000000000 ffffffff82399320 ffff88040897b598 ffffffff810dd54e [ 13.884130] ffff88040ac486a8 0000000000000000 0000000000000001 ffffffff82399320 [ 13.884132] Call Trace: [ 13.884134] [<ffffffff815ba622>] dump_stack+0x4c/0x65 [ 13.884137] [<ffffffff810dd54e>] validate_chain.isra.9+0x75d/0xf59 [ 13.884139] [<ffffffff815c62b3>] ? _raw_spin_unlock_irq+0x28/0x34 [ 13.884140] [<ffffffff810e0964>] __lock_acquire+0x753/0xabf [ 13.884142] [<ffffffff81163819>] ? kfree+0xb8/0x10d [ 13.884143] [<ffffffff810e1131>] lock_acquire+0x7b/0x9c [ 13.884146] [<ffffffffa0355dac>] ? snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884147] [<ffffffff815c4896>] down_read+0x44/0x8d [ 13.884150] [<ffffffffa0355dac>] ? snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884152] [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] [ 13.884153] [<ffffffff815c37e5>] ? __mutex_unlock_slowpath+0x110/0x11b [ 13.884156] [<ffffffffa0359338>] snd_usb_mixer_set_ctl_value+0x9a/0x16d [snd_usb_audio] [ 13.884159] [<ffffffffa035957e>] snd_usb_set_cur_mix_value+0x4d/0x77 [snd_usb_audio] [ 13.884161] [<ffffffffa0359db9>] restore_mixer_value+0x74/0x87 [snd_usb_audio] [ 13.884164] [<ffffffffa035b581>] snd_usb_mixer_resume+0x64/0x70 [snd_usb_audio] [ 13.884166] [<ffffffffa035505c>] __usb_audio_resume+0x5c/0xd8 [snd_usb_audio] [ 13.884169] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884171] [<ffffffffa03550e6>] usb_audio_reset_resume+0xe/0x10 [snd_usb_audio] [ 13.884172] [<ffffffff81412fde>] usb_resume_interface.isra.1+0x6c/0xbd [ 13.884174] [<ffffffff814130cf>] usb_resume_both+0xa0/0xc0 [ 13.884175] [<ffffffff81413cb9>] usb_runtime_resume+0x15/0x17 [ 13.884178] [<ffffffff813b92b0>] __rpm_callback+0x35/0x5d [ 13.884179] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884180] [<ffffffff813b9330>] rpm_callback+0x58/0x77 [ 13.884182] [<ffffffff81413ca4>] ? usb_runtime_suspend+0x63/0x63 [ 13.884183] [<ffffffff813ba0db>] rpm_resume+0x370/0x412 [ 13.884184] [<ffffffff813ba06c>] rpm_resume+0x301/0x412 [ 13.884186] [<ffffffff813ba1e3>] __pm_runtime_resume+0x66/0x82 [ 13.884187] [<ffffffff81412e10>] usb_autopm_get_interface+0x1e/0x48 [ 13.884190] [<ffffffffa0355dcc>] snd_usb_autoresume+0x3d/0x52 [snd_usb_audio] [ 13.884191] [<ffffffff810d84b9>] ? __init_waitqueue_head+0x37/0x4b [ 13.884194] [<ffffffffa035e765>] snd_usb_pcm_open+0x181/0x3bb [snd_usb_audio] [ 13.884197] [<ffffffffa035e9ad>] snd_usb_capture_open+0xe/0x10 [snd_usb_audio] [ 13.884200] [<ffffffffa02ac8c6>] snd_pcm_open_substream+0x50/0x9d [snd_pcm] [ 13.884203] [<ffffffffa02ac9ce>] snd_pcm_open+0xbb/0x1f8 [snd_pcm] [ 13.884204] [<ffffffff810de87d>] ? trace_hardirqs_on+0xd/0xf [ 13.884206] [<ffffffff810c5550>] ? wake_up_q+0x53/0x53 [ 13.884209] [<ffffffffa02acb49>] snd_pcm_capture_open+0x3e/0x61 [snd_pcm] [ 13.884212] [<ffffffffa028232d>] snd_open+0x130/0x13f [snd] [ 13.884214] [<ffffffff811788c5>] chrdev_open+0x13b/0x175 [ 13.884215] [<ffffffff8117878a>] ? cdev_put+0x20/0x20 [ 13.884217] [<ffffffff81172f31>] do_dentry_open.isra.1+0x1d2/0x2c9 [ 13.884218] [<ffffffff8117d0e4>] ? __inode_permission+0x84/0x95 [ 13.884219] [<ffffffff81173bf6>] vfs_open+0x50/0x54 [ 13.884221] [<ffffffff811815e0>] path_openat+0xa90/0xd0f [ 13.884222] [<ffffffff810e0900>] ? __lock_acquire+0x6ef/0xabf [ 13.884223] [<ffffffff8118269e>] do_filp_open+0x48/0xac [ 13.884225] [<ffffffff815c6203>] ? _raw_spin_unlock+0x23/0x2e [ 13.884227] [<ffffffff8118f5b7>] ? __alloc_fd+0x144/0x156 [ 13.884228] [<ffffffff81173f3a>] do_sys_open+0x154/0x1f7 [ 13.884229] [<ffffffff81173ff6>] SyS_open+0x19/0x1b [ 13.884231] [<ffffffff815c6a57>] entry_SYSCALL_64_fastpath+0x12/0x6f Previous version of the kernel was 4.2.0-rc6+.

3 11

[alsa-devel] [PATCH] ALSA: hda - Remove superfluous pcm NULL check
by Takashi Iwai 26 Aug '15

26 Aug '15

The variable pcm can be never NULL since it was rewritten with list_for_each_entry(). Suggested-by: Markus Osterhoff <linux-kernel(a)k-raum.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c021911f88a..37f43a1b34ef 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3102,8 +3102,7 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (!pcm || !pcm->pcm || pcm->own_chmap || - !hinfo->substreams) + if (!pcm->pcm || pcm->own_chmap || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, -- 2.5.0

1 0

[alsa-devel] [asoc:topic/rockchip 1/4] sound/soc/rockchip/rockchip_max98090.c:225:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 26 Aug '15

26 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f8ce20005d1694584448cd544be5fb32416a277c commit: 49bdb0440541ad1144ad08d5613f9a28bcd2a8dc [1/4] ASoC: rockchip: Add machine driver for max98090 codec coccinelle warnings: (new ones prefixed by >>) >> sound/soc/rockchip/rockchip_max98090.c:225:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [asoc:topic/rockchip 2/4] sound/soc/rockchip/rockchip_rt5645.c:214:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 26 Aug '15

26 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f8ce20005d1694584448cd544be5fb32416a277c commit: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 [2/4] ASoC: rockchip: Add machine driver for rt5645/rt5650 codec coccinelle warnings: (new ones prefixed by >>) >> sound/soc/rockchip/rockchip_rt5645.c:214:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH v4] ASoC: ics43432: Add codec driver for InvenSense ICS-43432 (fwd)
by Ricard Wanderlof 26 Aug '15

26 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Tested on a setup with a single ICS-43432, although the device itself supports stereo operation using a hardware pin controlling left vs. right channel output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V4: Updated commit message that the driver has been tested with real hardware, after off-list comment on V3. V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[alsa-devel] [PATCH 1/5] ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning
by Oder Chiou 26 Aug '15

26 Aug '15

Add i2c shutdown function to prevent the pop sound of the headphone while the system is rebooting or shutdowning Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e5fef8f..4189d5a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3490,13 +3490,23 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, 0x0020, 0x0020); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, 0x1000, 0x1000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, 0x0004, 0x0000); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); -- 1.8.1.1.439.g50a6b54

2 11

[alsa-devel] [PATCH v2] ASoC: rockchip: fix a misjudgement by return
by Xing Zheng 26 Aug '15

26 Aug '15

Being careless, judge the return value of snd_soc_card_jack_new is opposite, so it should be fixed. --- Changes in v2: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Reviewed-by: Dylan Reid <dgreid(a)chromium.org> sound/soc/rockchip/rockchip_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 3c6bb1e..adfe98c 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -118,7 +118,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, NULL, 0); - if (!ret) { + if (ret) { dev_err(card->dev, "New Headset Jack failed! (%d)\n", ret); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 26 Aug '15

26 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

5 11

[alsa-devel] [PATCH] ALSA: hda - Fix widget sysfs tree corruption after refresh
by Takashi Iwai 26 Aug '15

26 Aug '15

When snd_hdac_refresh_widget_sysfs() is called before the first hda_widget_sysfs_init(), the next call overrides and eventually fails. This results in unexpected Oops, something like: BUG: unable to handle kernel NULL pointer dereference at 00000000000000c8 IP: [<ffffffff8180e2a3>] hdmi_chmap_ctl_info+0x23/0x40 The fix is to add a check of the existing sysfs tree. Also, for more safety, this patch adds the checks of device_is_registered() in snd-hdac_refresh_wdiget_sysfs(), too. Fixes: fa4f18b4f402 ('ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs') Bugizlla: https://bugzilla.kernel.org/show_bug.cgi?id=103431 Reported-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 14 ++++++++------ sound/hda/hdac_sysfs.c | 3 +++ 2 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index aa6d6cec2380..db96042a497f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -384,18 +384,20 @@ int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec) { int ret; - hda_widget_sysfs_exit(codec); + if (device_is_registered(&codec->dev)) + hda_widget_sysfs_exit(codec); ret = snd_hdac_refresh_widgets(codec); if (ret) { dev_err(&codec->dev, "failed to refresh widget: %d\n", ret); return ret; } - ret = hda_widget_sysfs_init(codec); - if (ret) { - dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); - return ret; + if (device_is_registered(&codec->dev)) { + ret = hda_widget_sysfs_init(codec); + if (ret) { + dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); + return ret; + } } - return ret; } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widget_sysfs); diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 089b35f6f108..c71142dea98a 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -390,6 +390,9 @@ int hda_widget_sysfs_init(struct hdac_device *codec) { int err; + if (codec->widgets) + return 0; /* already created */ + err = widget_tree_create(codec); if (err < 0) { widget_tree_free(codec); -- 2.5.0

1 0

[alsa-devel] [asoc:topic/rt5677 6/6] sound/soc/codecs/rt5677-spi.c:130:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int'
by kbuild test robot 25 Aug '15

25 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5677 head: 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 commit: 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 [6/6] ASoC: rt5677: Allow arbitrary block read/write via SPI config: mn10300-allmodconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 7d4d443eb4386d6dbd420fa96303dd8fbc1eefc8 # save the attached .config to linux build tree make.cross ARCH=mn10300 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5677-spi.c: In function 'rt5677_spi_read': >> sound/soc/codecs/rt5677-spi.c:130:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat=] dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); ^ sound/soc/codecs/rt5677-spi.c: In function 'rt5677_spi_write': sound/soc/codecs/rt5677-spi.c:184:3: warning: format '%zu' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat=] dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); ^ vim +130 sound/soc/codecs/rt5677-spi.c 114 int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) 115 { 116 u32 offset; 117 int status = 0; 118 struct spi_transfer t[2]; 119 struct spi_message m; 120 /* +4 bytes is for the DummyPhase following the AddressPhase */ 121 u8 header[RT5677_SPI_HEADER + 4]; 122 u8 body[RT5677_SPI_BURST_LEN]; 123 u8 spi_cmd; 124 u8 *cb = rxbuf; 125 126 if (!g_spi) 127 return -ENODEV; 128 129 if ((addr & 1) || (len & 1)) { > 130 dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); 131 return -EACCES; 132 } 133 134 memset(t, 0, sizeof(t)); 135 t[0].tx_buf = header; 136 t[0].len = sizeof(header); 137 t[0].speed_hz = RT5677_SPI_FREQ; 138 t[1].rx_buf = body; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH V5 3/3] ASoC: AMD: add AMD ASoC ACP-I2S driver
by Alex Deucher 25 Aug '15

25 Aug '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP IP block consists of dedicated DMA and I2S blocks. The PCM driver provides the DMA and CPU DAI components to ALSA core. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Reviewed-by: Murali Krishna Vemuri <murali-krishna.vemuri(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: squash in Kconfig fix v3: squash additional commits, convert to mfd, drop rt286 changes v4: squash in naming fixes v5: move patch versioning out of commit message The module licensing follows the model used in the drm drivers. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 686 insertions(+) create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 2ae9619..1b9ce31 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/adi/Kconfig" +source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e189903..a6cbb99 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += adi/ +obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig new file mode 100644 index 0000000..ce39979 --- /dev/null +++ b/sound/soc/amd/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_AMD_ACP + tristate "AMD Audio Coprocessor support" + depends on MFD_CORE + help + This option enables ACP support (DMA,I2S) on AMD platforms. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile new file mode 100644 index 0000000..1a66ec0 --- /dev/null +++ b/sound/soc/amd/Makefile @@ -0,0 +1,3 @@ +snd-soc-acp-pcm-objs := acp-pcm-dma.o + +obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c new file mode 100644 index 0000000..18cc60c --- /dev/null +++ b/sound/soc/amd/acp-pcm-dma.c @@ -0,0 +1,676 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2014-2015 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR + * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, + * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + * + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/amd_acp.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 2 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 2 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 1024 + +#define NUM_DSCRS_PER_CHANNEL 2 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + + +static const struct snd_pcm_hardware acp_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + struct amd_acp_device *acp_dev; + struct acp_irq_prv *iprv; +}; + +static const struct snd_soc_component_driver dw_i2s_component = { + .name = "dw-i2s", +}; + +static void acp_pcm_period_elapsed(struct device *dev, u16 play_intr, + u16 capture_intr) +{ + struct snd_pcm_substream *substream; + struct audio_drv_data *irq_data = dev_get_drvdata(dev); + + /* Inform ALSA about the period elapsed (one out of two periods) */ + if (play_intr) + substream = irq_data->play_stream; + else + substream = irq_data->capture_stream; + + if (substream->runtime && snd_pcm_running(substream)) + snd_pcm_period_elapsed(substream); +} + +static int acp_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_drv_data *intr_data = dev_get_drvdata(prtd->platform->dev); + + struct audio_substream_data *adata = + kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); + if (adata == NULL) + return -ENOMEM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp_pcm_hardware_playback; + else + runtime->hw = acp_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(prtd->platform->dev, "set integer constraint failed\n"); + return ret; + } + + adata->acp_dev = intr_data->acp_dev; + runtime->private_data = adata; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + intr_data->play_stream = substream; + else + intr_data->capture_stream = substream; + + return 0; +} + +static int acp_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + uint64_t size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + u16 num_of_pages; + struct snd_pcm_runtime *runtime; + struct audio_substream_data *rtd; + struct amd_acp_device *acp_dev; + + dma_buffer = &substream->dma_buffer; + + runtime = substream->runtime; + rtd = runtime->private_data; + + if (WARN_ON(!rtd)) + return -EINVAL; + acp_dev = rtd->acp_dev; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + + if (NULL != pg) { + /* Save for runtime private data */ + rtd->pg = pg; + rtd->order = get_order(size); + + /* Let ACP know the Allocated memory */ + num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; + + /* Fill the page table entries in ACP SRAM */ + rtd->pg = pg; + rtd->size = size; + rtd->num_of_pages = num_of_pages; + rtd->direction = substream->stream; + + acp_dev->config_dma(acp_dev, rtd); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static int acp_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + pos = acp_dev->update_dma_pointer(acp_dev, substream->stream, + frames_to_bytes(runtime, runtime->period_size)); + return bytes_to_frames(runtime, pos); + +} + +static int acp_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp_dma_prepare(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, ACP_TO_I2S_DMA_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH13, + NUM_DSCRS_PER_CHANNEL, 0); + /* Fill ACP SRAM (2 periods) with zeros from System RAM + * which is zero-ed in hw_params */ + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + ret = -EFAULT; + + /* Now configure DMA to transfer only first half of System RAM + * buffer before playback is triggered. This will overwrite + * zero-ed second half of SRAM buffer */ + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + 1, 0); + } else { + acp_dev->config_dma_channel(acp_dev, ACP_TO_SYSRAM_CH_NUM, + CAPTURE_START_DMA_DESCR_CH14, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, I2S_TO_ACP_DMA_CH_NUM, + CAPTURE_START_DMA_DESCR_CH15, + NUM_DSCRS_PER_CHANNEL, 0); + } + return 0; +} + +static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + if (!rtd) + return -EINVAL; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + return -EFAULT; + acp_dev->prebuffer_audio(rtd->acp_dev); + + ret = acp_dev->dma_start(acp_dev, + ACP_TO_I2S_DMA_CH_NUM, true); + } else { + ret = acp_dev->dma_start(acp_dev, + I2S_TO_ACP_DMA_CH_NUM, true); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_stop(acp_dev, SYSRAM_TO_ACP_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_I2S_DMA_CH_NUM); + } else { + ret = acp_dev->dma_stop(acp_dev, I2S_TO_ACP_DMA_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_SYSRAM_CH_NUM); + } + break; + default: + ret = -EINVAL; + + } + return ret; +} + +static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_pcm *pcm; + + pcm = rtd->pcm; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, MAX_BUFFER); + return ret; +} + +static int acp_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + + kfree(rtd); + + pm_runtime_mark_last_busy(prtd->platform->dev); + return 0; +} + +static int acp_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + struct device *dev = prtd->platform->dev; + u32 chan_nr; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + + default: + dev_err(dev, "unsuppted PCM fmt : %d\n", params_format(params)); + return -EINVAL; + } + + chan_nr = params_channels(params); + + switch (chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + rtd->ch_reg = 3; + break; + case SIX_CHANNEL_SUPPORT: + rtd->ch_reg = 2; + break; + case FOUR_CHANNEL_SUPPORT: + rtd->ch_reg = 1; + break; + case TWO_CHANNEL_SUPPORT: + rtd->ch_reg = 0; + break; + default: + dev_err(dev, "channel not supported : %d\n", chan_nr); + return -EINVAL; + } + + rtd->direction = substream->stream; + + acp_dev->config_i2s(acp_dev, rtd); + + return 0; +} + +static int acp_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + acp_dev->i2s_start(acp_dev, substream->stream); + ret = 0; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + acp_dev->i2s_stop(acp_dev, substream->stream); + ret = 0; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int acp_dai_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + acp_dev->i2s_reset(acp_dev, substream->stream); + return 0; +} + +static struct snd_pcm_ops acp_dma_ops = { + .open = acp_dma_open, + .close = acp_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp_dma_hw_params, + .hw_free = acp_dma_hw_free, + .trigger = acp_dma_trigger, + .pointer = acp_dma_pointer, + .mmap = acp_dma_mmap, + .prepare = acp_dma_prepare, +}; + +static struct snd_soc_dai_ops acp_dai_i2s_ops = { + .prepare = acp_dai_i2s_prepare, + .hw_params = acp_dai_i2s_hwparams, + .trigger = acp_dai_i2s_trigger, +}; + +static struct snd_soc_dai_driver i2s_dai_driver = { + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp_dai_i2s_ops, +}; + +static struct snd_soc_platform_driver acp_asoc_platform = { + .ops = &acp_dma_ops, + .pcm_new = acp_dma_new, +}; + +static int acp_alsa_register(struct device *dev, struct amd_acp_device *acp_dev, + struct platform_device *pdev) +{ + int status; + + status = snd_soc_register_platform(dev, &acp_asoc_platform); + if (0 != status) { + dev_err(dev, "Unable to register ALSA platform device\n"); + goto exit_platform; + } else { + status = snd_soc_register_component(dev, + &dw_i2s_component, + &i2s_dai_driver, 1); + + if (0 != status) { + dev_err(dev, "Unable to register i2s dai\n"); + goto exit_dai; + } else { + dev_info(dev, "ACP device registered with ALSA\n"); + return status; + } + } + +exit_dai: + snd_soc_unregister_platform(dev); +exit_platform: + acp_dev->fini(acp_dev); + return status; +} + +static int acp_audio_probe(struct platform_device *pdev) +{ + int status; + struct audio_drv_data *audio_drv_data; + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), + GFP_KERNEL); + if (audio_drv_data == NULL) + return -ENOMEM; + + audio_drv_data->iprv = devm_kzalloc(&pdev->dev, + sizeof(struct acp_irq_prv), + GFP_KERNEL); + if (audio_drv_data->iprv == NULL) + return -ENOMEM; + + /* The following members gets populated in device 'open' + * function. Till then interrupts are disabled in 'acp_hw_init' + * and device doesn't generate any interrupts. + */ + + audio_drv_data->play_stream = NULL; + audio_drv_data->capture_stream = NULL; + audio_drv_data->acp_dev = acp_dev; + + audio_drv_data->iprv->dev = &pdev->dev; + audio_drv_data->iprv->acp_dev = acp_dev; + audio_drv_data->iprv->set_elapsed = acp_pcm_period_elapsed; + + dev_set_drvdata(&pdev->dev, audio_drv_data); + + /* Initialize the ACP */ + status = acp_dev->init(acp_dev, audio_drv_data->iprv); + + if (0 == status) + status = acp_alsa_register(&pdev->dev, acp_dev, pdev); + else + dev_err(&pdev->dev, "ACP initialization Failed\n"); + + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); + + return status; +} + +static int acp_audio_remove(struct platform_device *pdev) +{ + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + + acp_dev->fini(acp_dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static int acp_pcm_suspend(struct device *dev) +{ + bool pm_rts; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == false) + adata->acp_dev->fini(adata->acp_dev); + + return 0; +} + +static int acp_pcm_resume(struct device *dev) +{ + bool pm_rts; + struct snd_pcm_substream *pstream, *cstream; + struct snd_pcm_runtime *prtd, *crtd; + struct audio_substream_data *rtd; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == true) { + /* Resumed from system wide suspend and there is + * no pending audio activity to resume. */ + pm_runtime_disable(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + goto out; + } + + pstream = adata->play_stream; + prtd = pstream ? pstream->runtime : NULL; + if (prtd != NULL) { + /* Resume playback stream from a suspended state */ + rtd = prtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } + + cstream = adata->capture_stream; + crtd = cstream ? cstream->runtime : NULL; + if (crtd != NULL) { + /* Resume capture stream from a suspended state */ + rtd = crtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } +out: + return 0; +} + +static int acp_pcm_runtime_suspend(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_suspend(adata->acp_dev); + return 0; +} + +static int acp_pcm_runtime_resume(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_resume(adata->acp_dev); + return 0; +} + +static const struct dev_pm_ops acp_pm_ops = { + .suspend = acp_pcm_suspend, + .resume = acp_pcm_resume, + .runtime_suspend = acp_pcm_runtime_suspend, + .runtime_resume = acp_pcm_runtime_resume, +}; + +static struct platform_driver acp_dma_driver = { + .probe = acp_audio_probe, + .remove = acp_audio_remove, + .driver = { + .name = "acp-i2s-audio", + .pm = &acp_pm_ops, + }, +}; + +module_platform_driver(acp_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP PCM Driver"); +MODULE_LICENSE("GPL and additional rights"); +MODULE_ALIAS("platform:acp-i2s-audio"); -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: rt5645: Add struct dmi_system_id "Google Celes" for chrome platform
by Oder Chiou 25 Aug '15

25 Aug '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4917057..bae1c5f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3198,6 +3198,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Strago"), }, }, + { + .ident = "Google Celes", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: arizona: Poll for FLL clock OK rather than use interrupts
by Charles Keepax 25 Aug '15

25 Aug '15

The extcon driver takes the DAPM mutex from within the interrupt thread in several places, which makes it possible to get into a situation where the interrupt thread is blocked waiting on the DAPM mutex whilst a DAPM sequence is running which is attempting to configure the FLL. In this case the FLL completion can't be completed as as the IRQ handler is ONE_SHOT, which cause the FLL lock to use the full time out (250mS) and report that the process timed out. It is not really practical to make the extcon driver not take the DAPM mutex from within the interrupt thread, at least not without extensive modification. So this patch fixes the issue by switching the wait for the FLL lock to polling. A few fast polls are done first as the FLL should lock quickly for a good quality reference clock, (indeed it hits on the first poll on my system) and it will poll every 20mS after that until it times out. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 47 +++++++++++++++++-------------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 19 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 42b2021..8a2221a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1756,17 +1756,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_clock_ok(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "clock OK\n"); - - complete(&fll->ok); - - return IRQ_HANDLED; -} - static struct { unsigned int min; unsigned int max; @@ -2048,10 +2037,11 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + int i; + unsigned int val; if (already_enabled < 0) return already_enabled; @@ -2110,9 +2100,6 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (!already_enabled) pm_runtime_get(arizona->dev); - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) @@ -2124,10 +2111,24 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - time_left = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (time_left == 0) + arizona_fll_dbg(fll, "Waiting for FLL lock...\n"); + val = 0; + for (i = 0; i < 15; i++) { + if (i < 5) + usleep_range(200, 400); + else + msleep(20); + + regmap_read(arizona->regmap, + ARIZONA_INTERRUPT_RAW_STATUS_5, + &val); + if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1))) + break; + } + if (i == 15) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + else + arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i); return 0; } @@ -2212,11 +2213,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll); int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { - int ret; unsigned int val; - init_completion(&fll->ok); - fll->id = id; fll->base = base; fll->arizona = arizona; @@ -2238,13 +2236,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, - arizona_fll_clock_ok, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", - id, ret); - } - regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 13598b0..ada0a41 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -261,7 +261,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion ok; unsigned int fout; int sync_src; -- 1.7.2.5

2 1

[alsa-devel] [PATCH v1 0/1] A misjudgement should be fixed
by Xing Zheng 25 Aug '15

25 Aug '15

Hi, Being careless, judge the return value of snd_soc_card_jack_new is opposite, so it should be fixed. Thanks. Changes in v1: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Reviewed-by: Dylan Reid <dgreid(a)chromium.org> Xing Zheng (1): ASoC: rockchip: fix a misjudgement by return sound/soc/rockchip/rockchip_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: rt5677: Allow arbitrary block read/write via SPI
by Ben Zhang 25 Aug '15

25 Aug '15

Added rt5677_spi_read() and refactored rt5677_spi_write() so that an arbitrary block in the DSP address space can be read/written via SPI. For example, this allows us to load an ELF DSP firmware with sparse sections, and stream audio samples from DSP ring buffer. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 233 +++++++++++++++++++++++++++++++----------- sound/soc/codecs/rt5677-spi.h | 8 +- sound/soc/codecs/rt5677.c | 4 +- 3 files changed, 178 insertions(+), 67 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index ef6348c..3505aaf 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -31,84 +31,197 @@ #include "rt5677-spi.h" +#define RT5677_SPI_BURST_LEN 240 +#define RT5677_SPI_HEADER 5 +#define RT5677_SPI_FREQ 6000000 + +/* The AddressPhase and DataPhase of SPI commands are MSB first on the wire. + * DataPhase word size of 16-bit commands is 2 bytes. + * DataPhase word size of 32-bit commands is 4 bytes. + * DataPhase word size of burst commands is 8 bytes. + * The DSP CPU is little-endian. + */ +#define RT5677_SPI_WRITE_BURST 0x5 +#define RT5677_SPI_READ_BURST 0x4 +#define RT5677_SPI_WRITE_32 0x3 +#define RT5677_SPI_READ_32 0x2 +#define RT5677_SPI_WRITE_16 0x1 +#define RT5677_SPI_READ_16 0x0 + static struct spi_device *g_spi; +static DEFINE_MUTEX(spi_mutex); -/** - * rt5677_spi_write - Write data to SPI. - * @txbuf: Data Buffer for writing. - * @len: Data length. +/* Select a suitable transfer command for the next transfer to ensure + * the transfer address is always naturally aligned while minimizing + * the total number of transfers required. + * + * 3 transfer commands are available: + * RT5677_SPI_READ/WRITE_16: Transfer 2 bytes + * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes + * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes + * + * For example, reading 260 bytes at 0x60030002 uses the following commands: + * 0x60030002 RT5677_SPI_READ_16 2 bytes + * 0x60030004 RT5677_SPI_READ_32 4 bytes + * 0x60030008 RT5677_SPI_READ_BURST 240 bytes + * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes + * 0x60030100 RT5677_SPI_READ_32 4 bytes + * 0x60030104 RT5677_SPI_READ_16 2 bytes * + * Input: + * @read: true for read commands; false for write commands + * @align: alignment of the next transfer address + * @remain: number of bytes remaining to transfer * - * Returns true for success. + * Output: + * @len: number of bytes to transfer with the selected command + * Returns the selected command */ -int rt5677_spi_write(u8 *txbuf, size_t len) +static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { - int status; - - status = spi_write(g_spi, txbuf, len); - - if (status) - dev_err(&g_spi->dev, "rt5677_spi_write error %d\n", status); - - return status; + u8 cmd; + + if (align == 2 || align == 6 || remain == 2) { + cmd = RT5677_SPI_READ_16; + *len = 2; + } else if (align == 4 || remain <= 6) { + cmd = RT5677_SPI_READ_32; + *len = 4; + } else { + cmd = RT5677_SPI_READ_BURST; + *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + } + return read ? cmd : cmd + 1; } -EXPORT_SYMBOL_GPL(rt5677_spi_write); -/** - * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. - * @addr: Start address. - * @txbuf: Data Buffer for writng. - * @len: Data length, it must be a multiple of 8. - * - * - * Returns true for success. +/* Copy dstlen bytes from src to dst, while reversing byte order for each word. + * If srclen < dstlen, zeros are padded. */ -int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) +static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) { - u8 spi_cmd = RT5677_SPI_CMD_BURST_WRITE; - u8 *write_buf; - unsigned int i, end, offset = 0; - - write_buf = kmalloc(RT5677_SPI_BUF_LEN + 6, GFP_KERNEL); - - if (write_buf == NULL) - return -ENOMEM; - - while (offset < fw->size) { - if (offset + RT5677_SPI_BUF_LEN <= fw->size) - end = RT5677_SPI_BUF_LEN; - else - end = fw->size % RT5677_SPI_BUF_LEN; - - write_buf[0] = spi_cmd; - write_buf[1] = ((addr + offset) & 0xff000000) >> 24; - write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; - write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; - write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; - - for (i = 0; i < end; i += 8) { - write_buf[i + 12] = fw->data[offset + i + 0]; - write_buf[i + 11] = fw->data[offset + i + 1]; - write_buf[i + 10] = fw->data[offset + i + 2]; - write_buf[i + 9] = fw->data[offset + i + 3]; - write_buf[i + 8] = fw->data[offset + i + 4]; - write_buf[i + 7] = fw->data[offset + i + 5]; - write_buf[i + 6] = fw->data[offset + i + 6]; - write_buf[i + 5] = fw->data[offset + i + 7]; + u32 w, i, si; + u32 word_size = min_t(u32, dstlen, 8); + + for (w = 0; w < dstlen; w += word_size) { + for (i = 0; i < word_size; i++) { + si = w + word_size - i - 1; + dst[w + i] = si < srclen ? src[si] : 0; } + } +} - write_buf[end + 5] = spi_cmd; +/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) +{ + u32 offset; + int status = 0; + struct spi_transfer t[2]; + struct spi_message m; + /* +4 bytes is for the DummyPhase following the AddressPhase */ + u8 header[RT5677_SPI_HEADER + 4]; + u8 body[RT5677_SPI_BURST_LEN]; + u8 spi_cmd; + u8 *cb = rxbuf; + + if (!g_spi) + return -ENODEV; + + if ((addr & 1) || (len & 1)) { + dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); + return -EACCES; + } - rt5677_spi_write(write_buf, end + 6); + memset(t, 0, sizeof(t)); + t[0].tx_buf = header; + t[0].len = sizeof(header); + t[0].speed_hz = RT5677_SPI_FREQ; + t[1].rx_buf = body; + t[1].speed_hz = RT5677_SPI_FREQ; + spi_message_init_with_transfers(&m, t, ARRAY_SIZE(t)); + + for (offset = 0; offset < len; offset += t[1].len) { + spi_cmd = rt5677_spi_select_cmd(true, (addr + offset) & 7, + len - offset, &t[1].len); + + /* Construct SPI message header */ + header[0] = spi_cmd; + header[1] = ((addr + offset) & 0xff000000) >> 24; + header[2] = ((addr + offset) & 0x00ff0000) >> 16; + header[3] = ((addr + offset) & 0x0000ff00) >> 8; + header[4] = ((addr + offset) & 0x000000ff) >> 0; + + mutex_lock(&spi_mutex); + status |= spi_sync(g_spi, &m); + mutex_unlock(&spi_mutex); + + /* Copy data back to caller buffer */ + rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + } + return status; +} +EXPORT_SYMBOL_GPL(rt5677_spi_read); - offset += RT5677_SPI_BUF_LEN; +/* Write DSP address space using SPI. addr has to be 2-byte aligned. + * If len is not 2-byte aligned, an extra byte of zero is written at the end + * as padding. + */ +int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) +{ + u32 offset, len_with_pad = len; + int status = 0; + struct spi_transfer t; + struct spi_message m; + /* +1 byte is for the DummyPhase following the DataPhase */ + u8 buf[RT5677_SPI_HEADER + RT5677_SPI_BURST_LEN + 1]; + u8 *body = buf + RT5677_SPI_HEADER; + u8 spi_cmd; + const u8 *cb = txbuf; + + if (!g_spi) + return -ENODEV; + + if (addr & 1) { + dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); + return -EACCES; } - kfree(write_buf); + if (len & 1) + len_with_pad = len + 1; + + memset(&t, 0, sizeof(t)); + t.tx_buf = buf; + t.speed_hz = RT5677_SPI_FREQ; + spi_message_init_with_transfers(&m, &t, 1); + + for (offset = 0; offset < len_with_pad;) { + spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, + len_with_pad - offset, &t.len); + + /* Construct SPI message header */ + buf[0] = spi_cmd; + buf[1] = ((addr + offset) & 0xff000000) >> 24; + buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + /* Fetch data from caller buffer */ + rt5677_spi_reverse(body, t.len, cb + offset, len - offset); + offset += t.len; + t.len += RT5677_SPI_HEADER + 1; + + mutex_lock(&spi_mutex); + status |= spi_sync(g_spi, &m); + mutex_unlock(&spi_mutex); + } + return status; +} +EXPORT_SYMBOL_GPL(rt5677_spi_write); - return 0; +int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw) +{ + return rt5677_spi_write(addr, fw->data, fw->size); } -EXPORT_SYMBOL_GPL(rt5677_spi_burst_write); +EXPORT_SYMBOL_GPL(rt5677_spi_write_firmware); static int rt5677_spi_probe(struct spi_device *spi) { diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index ec41b2b..662db16 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -12,10 +12,8 @@ #ifndef __RT5677_SPI_H__ #define __RT5677_SPI_H__ -#define RT5677_SPI_BUF_LEN 240 -#define RT5677_SPI_CMD_BURST_WRITE 0x05 - -int rt5677_spi_write(u8 *txbuf, size_t len); -int rt5677_spi_burst_write(u32 addr, const struct firmware *fw); +int rt5677_spi_read(u32 addr, void *rxbuf, size_t len); +int rt5677_spi_write(u32 addr, const void *txbuf, size_t len); +int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw); #endif /* __RT5677_SPI_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3f890a6..d916d0c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -745,14 +745,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); if (ret == 0) { - rt5677_spi_burst_write(0x50000000, rt5677->fw1); + rt5677_spi_write_firmware(0x50000000, rt5677->fw1); release_firmware(rt5677->fw1); } ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2, codec->dev); if (ret == 0) { - rt5677_spi_burst_write(0x60000000, rt5677->fw2); + rt5677_spi_write_firmware(0x60000000, rt5677->fw2); release_firmware(rt5677->fw2); } -- 2.5.0.457.gab17608

3 2

Re: [alsa-devel] [RESEND PATCH v1 0/1] A misjudgement should be fixed
by Mark Brown 25 Aug '15

25 Aug '15

On Tue, Aug 25, 2015 at 03:52:41PM +0800, Xing Zheng wrote: > > Hi, > Being careless, judge the return value of snd_soc_card_jack_new > is opposite, so it should be fixed. Please don't send cover letters for single patches, if there is anything useful it should either be in the changelog or after the --- in the patch.

1 0

[alsa-devel] [sound:for-next 174/174] sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE'
by kbuild test robot 25 Aug '15

25 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: a2bc2af66a5b656577fc2d71b6c277b73537777e commit: a2bc2af66a5b656577fc2d71b6c277b73537777e [174/174] ALSA: ppc: keywest: Export I2C module alias information config: powerpc-defconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout a2bc2af66a5b656577fc2d71b6c277b73537777e # save the attached .config to linux build tree make.cross ARCH=powerpc All error/warnings (new ones prefixed by >>): >> sound/ppc/keywest.c:104:1: warning: data definition has no type or storage class MODULE_DEVICE_TABLE(i2c, keywest_i2c_id); ^ >> sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] >> sound/ppc/keywest.c:104:1: warning: parameter names (without types) in function declaration cc1: some warnings being treated as errors vim +104 sound/ppc/keywest.c 98 99 static const struct i2c_device_id keywest_i2c_id[] = { 100 { "MAC,tas3004", 0 }, /* instantiated by i2c-powermac */ 101 { "keywest", 0 }, /* instantiated by us if needed */ 102 { } 103 }; > 104 MODULE_DEVICE_TABLE(i2c, keywest_i2c_id); 105 106 static struct i2c_driver keywest_driver = { 107 .driver = { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 3

[alsa-devel] [PATCH] ALSA: ppc: Add missing inclusion of linux/module.h
by Takashi Iwai 25 Aug '15

25 Aug '15

Otherwise it triggers a compile warning like: sound/ppc/keywest.c:104:1: warning: data definition has no type or storage class sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] Fixes: a2bc2af66a5b ('ALSA: ppc: keywest: Export I2C module alias information') Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index f644a8c57e0a..4373615f13e2 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -22,6 +22,7 @@ #include <linux/init.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/module.h> #include <sound/core.h> #include "pmac.h" -- 2.5.0

2 1

[alsa-devel] [RESEND PATCH 0/7] Export I2C and OF module aliases in missing drivers
by Javier Martinez Canillas 25 Aug '15

25 Aug '15

Hello, This is a resend of the patches that were not picked from the series "[PATCH 00/27] Export I2C and OF module aliases in missing drivers" [0] posted about a month ago. The patches have no dependencies and can be picked individually by the relevant maintainer. I preferred to resend instead of sending a naked ping for each patch that I got no answer. Best regards, Javier Javier Martinez Canillas (7): i2c: core: Export I2C module alias information in dummy driver backlight: tosa: Export I2C module alias information usb: phy: isp1301: Export I2C module alias information ALSA: ppc: keywest: Export I2C module alias information extcon: Export OF module alias information in missing drivers leds: Export OF module alias information in missing drivers regulator: isl9305: Export OF module alias information drivers/extcon/extcon-rt8973a.c | 1 + drivers/extcon/extcon-sm5502.c | 1 + drivers/i2c/i2c-core.c | 1 + drivers/leds/leds-pca963x.c | 1 + drivers/leds/leds-tca6507.c | 1 + drivers/regulator/isl9305.c | 1 + drivers/usb/phy/phy-isp1301.c | 1 + drivers/video/backlight/tosa_bl.c | 1 + sound/ppc/keywest.c | 1 + 9 files changed, 9 insertions(+) -- 2.4.3

2 2

[alsa-devel] [PATCH 1/3] ASoC: davinci: Convert to use devm_ioremap_resource
by Axel Lin 25 Aug '15

25 Aug '15

Use devm_ioremap_resource() instead of open code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/davinci/davinci-i2s.c | 25 ++++++------------------- sound/soc/davinci/davinci-mcasp.c | 18 ++++-------------- 2 files changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 56cb4d9..ec98548 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -651,23 +651,15 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea, *res; + struct resource *mem, *res; + void __iomem *io_base; int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; - } - - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), - pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "McBSP region already claimed\n"); - return -EBUSY; - } + io_base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); @@ -679,12 +671,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; clk_enable(dev->clk); - dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!dev->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release_clk; - } + dev->base = io_base; dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b960e62..add6bb9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1613,7 +1613,7 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; - struct resource *mem, *ioarea, *res, *dat; + struct resource *mem, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; @@ -1648,22 +1648,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } } - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "Audio region already claimed\n"); - return -EBUSY; - } + mcasp->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(mcasp->base)) + return PTR_ERR(mcasp->base); pm_runtime_enable(&pdev->dev); - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!mcasp->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { -- 2.1.0

3 7

[alsa-devel] Applied "ASoC: cs4349: fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 25 Aug '15

25 Aug '15

The patch ASoC: cs4349: fix platform_no_drv_owner.cocci warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f755d114cba98708932cba89d8c98c4ab89a83b0 Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Sun, 23 Aug 2015 00:30:19 +0800 Subject: [PATCH] ASoC: cs4349: fix platform_no_drv_owner.cocci warnings sound/soc/codecs/cs4349.c:389:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 0d010c2b6021..0ac8fc5ed4ae 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -378,7 +378,6 @@ MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", - .owner = THIS_MODULE, .of_match_table = cs4349_of_match, }, .id_table = cs4349_i2c_id, -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: tegra: Fix unused variable 'spdif' warning
by Axel Lin 25 Aug '15

25 Aug '15

Fix below build warning: CC [M] sound/soc/tegra/tegra20_spdif.o sound/soc/tegra/tegra20_spdif.c: In function 'tegra20_spdif_platform_remove': sound/soc/tegra/tegra20_spdif.c:361:24: warning: unused variable 'spdif' [-Wunused-variable] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/tegra/tegra20_spdif.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 66237bc..0809b1e 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -358,8 +358,6 @@ err_pm_disable: static int tegra20_spdif_platform_remove(struct platform_device *pdev) { - struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); -- 2.1.0

3 2

[alsa-devel] [PATCH] ALSA: hda - Refresh sysfs at snd_hda_codec_update_widgets()
by Takashi Iwai 25 Aug '15

25 Aug '15

In the commit [fa4f18b4f402: ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs], snd_hdac_refresh_widget_sysfs() is explicitly called in the codec driver. But this results in refreshing twice, as snd_hdac_refresh_widget_sysfs() itself calls snd_hdac_refresh_widgets() function. Instead, we can replace the call in snd_hda_codec_update_widgets() with snd_hdac_refresh_widget_sysfs(). This also fixes the missing sysfs update for ca0132, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 792948418cf0..2c021911f88a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -955,7 +955,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; - err = snd_hdac_refresh_widgets(&codec->core); + err = snd_hdac_refresh_widget_sysfs(&codec->core); if (err < 0) return err; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index df982fc6a872..a97db5fc8a15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2331,7 +2331,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (is_haswell_plus(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); - snd_hdac_refresh_widget_sysfs(&codec->core); } /* For Valleyview/Cherryview, only the display codec is in the display -- 2.5.0

1 0

[alsa-devel] [PATCH v5 0/6] Add dw_hdmi i2s audio support
by Yakir Yang 25 Aug '15

25 Aug '15

As Russell have send an series patches for dw_hdmi audio, and most important it works well on rockchip platform. Cause some patches have been achieve in Russell's series, so I abondon some patches in this verison. Here are the abondon patches that already be achieved: * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks And I also abondon some interrupt patches that is not a common bug in dw_hdmi driver, but is a rockchip platform bug which HDMI registers would reset by CRU module after system suspend/resume (in this case HPD interrupt mask would reset, then driver can't respond hot plug event after system resume). Here are the abondon patches that used to "fix" dw_hdmi interrupt bug: * drm: bridge/dw_hdmi: add irq control to suspend/resume * drm: bridge/dw_hdmi: wrap irq control in fucntions * drm: rockchip/dw_hdmi_rockchip: add resume/suspend support Here are the leave list that I have make in my series: - add n/cts combinations for more display modes - enable audio support when sink device is HDMI monitor and has audio - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver - add rockchip_hdmi_audio sound card driver - add dt-bindings for documentation for rockchip_hdmi_audio driver Changes in v5: - make more words in commit message. - Take Russell suggest that create "sink_has_audio" and "sink_is_hdmi" in struct hdmi, and keep vmode's mdvi changeless. - Config fc_invidconf to HDMI mode when sink device is HDMI monitor. - Take Mark Brown suggest that remove jack_status recorded, report jack status directly when hdmi plug happend, and remove devm_kfree and devm_free_irq. - Correct the MODULE_LICENSE to "GPL v2" - Move source from sound/soc/codecs to drivers/gpu/drm/bridge/ - Rename dai driver name to "dw-hdmi-i2s-hifi" - Take Mark Brown suggest that remove the no useful code in probe func, and remove the snd_soc_dai_set_fmt() snd_soc_unregister_card(), remove those noisy dev_info() - Rename codec_dai_name to "dw-hdmi-i2s-hifi" Changes in v4: - Take Dainel suggest that introduce drm_detect_monitor_audio() to judge whether source should config audio. - Replace delaywork with irq thread, and add suspend/resume interfaces, replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (6): drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio when sink device is HDMI and has audio drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for rockchip_hdmi_audio driver .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 398 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 100 +++++- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 138 +++++++ 11 files changed, 658 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

4 10

[alsa-devel] [PATCH 1/2] ALSA: hda - Check all inputs for is_active_nid_for_any()
by Takashi Iwai 25 Aug '15

25 Aug '15

The is_active_nid_for_any() function in the generic parser is supposed to check all connections from/to the given widget, but the current code checks only the first input connection (index = 0). This patch corrects the code to check all inputs by passing -1 to index argument. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102521 Cc: <stable(a)vger.kernel.org> [v4.1+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b077bb644434..368181d8b0ce 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -671,7 +671,8 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, } for (i = 0; i < path->depth; i++) { if (path->path[i] == nid) { - if (dir == HDA_OUTPUT || path->idx[i] == idx) + if (dir == HDA_OUTPUT || idx == -1 || + path->idx[i] == idx) return true; break; } @@ -682,7 +683,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, /* check whether the NID is referred by any active paths */ #define is_active_nid_for_any(codec, nid) \ - is_active_nid(codec, nid, HDA_OUTPUT, 0) + is_active_nid(codec, nid, HDA_OUTPUT, -1) /* get the default amp value for the target state */ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, -- 2.5.0

1 1

[alsa-devel] [PATCH (alsa-lib)] pcm: Fix doxygen for two enums
by David Henningsson 24 Aug '15

24 Aug '15

The doxygen comments were wrong, making doxygen output weird. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/pcm.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index a1d14a9..0be1a32 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -324,9 +324,9 @@ typedef enum _snd_pcm_tstamp { } snd_pcm_tstamp_t; typedef enum _snd_pcm_tstamp_type { - SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ + SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /**< gettimeofday equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /**< posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /**< monotonic_raw (no NTP) */ SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; @@ -527,11 +527,11 @@ int snd_pcm_unlink(snd_pcm_t *pcm); /** channel map list type */ enum snd_pcm_chmap_type { - SND_CHMAP_TYPE_NONE = 0,/** unspecified channel position */ - SND_CHMAP_TYPE_FIXED, /** fixed channel position */ - SND_CHMAP_TYPE_VAR, /** freely swappable channel position */ - SND_CHMAP_TYPE_PAIRED, /** pair-wise swappable channel position */ - SND_CHMAP_TYPE_LAST = SND_CHMAP_TYPE_PAIRED, /** last entry */ + SND_CHMAP_TYPE_NONE = 0,/**< unspecified channel position */ + SND_CHMAP_TYPE_FIXED, /**< fixed channel position */ + SND_CHMAP_TYPE_VAR, /**< freely swappable channel position */ + SND_CHMAP_TYPE_PAIRED, /**< pair-wise swappable channel position */ + SND_CHMAP_TYPE_LAST = SND_CHMAP_TYPE_PAIRED, /**< last entry */ }; /** channel positions */ -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] [resend #2] snd-waveartist: Introduce Rockwell WaveArtist RWA010 driver
by Ondrej Zary 24 Aug '15

24 Aug '15

This is a new ALSA driver driver for Rockwell WaveArtist RWA010 chips found on some (rare) ISA sound cards, such as DCS Multimedia S717. I wasn't able to get the old OSS WaveArtist driver to work with my card but it was a great source of information about the chip (as the full datasheet is not available - only a brief one and a design guide). However, the OSS driver only supports few of the mixer registers so I had to install the card in Windows and dump mixer registers while changing the mixer settings. Then tried what the rest of the registers do and the result is a fully working mixer which even supports more controls than the Windows driver. Someone with a NetWinder can add support for it to this driver and then remove the old OSS one. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/mpu401.h | 1 + sound/isa/Kconfig | 11 + sound/isa/Makefile | 2 + sound/isa/waveartist.c | 1399 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/waveartist.h | 74 +++ 5 files changed, 1487 insertions(+) create mode 100644 sound/isa/waveartist.c create mode 100644 sound/isa/waveartist.h diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index e942096..8fceeba 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -44,6 +44,7 @@ #define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */ #define MPU401_HW_PC98II 18 /* Roland PC98II */ #define MPU401_HW_AUREAL 19 /* Aureal Vortex */ +#define MPU401_HW_WAVEARTIST 20 /* Rockwell WaveArtist */ #define MPU401_INFO_INPUT (1 << 0) /* input stream */ #define MPU401_INFO_OUTPUT (1 << 1) /* output stream */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..7a3b4a2 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -454,5 +454,16 @@ config SND_MSND_CLASSIC To compile this driver as a module, choose M here: the module will be called snd-msnd-classic. +config SND_WAVEARTIST + tristate "Rockwell WaveArtist RWA010" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + help + Say Y here to include support for Rockwell WaveArtist RWA010 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-waveartist. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..23e11e2 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -12,6 +12,7 @@ snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o snd-sscape-objs := sscape.o +snd-waveartist-objs := waveartist.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o +obj-$(CONFIG_SND_WAVEARTIST) += snd-waveartist.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/waveartist.c b/sound/isa/waveartist.c new file mode 100644 index 0000000..e5bda5c --- /dev/null +++ b/sound/isa/waveartist.c @@ -0,0 +1,1399 @@ +/* + * Driver for Rockwell WaveArtist RWA010 soundcards + * + * Copyright (c) 2015 Ondrej Zary + * + * HW-related parts based on OSS WaveArtist driver by Hannu Savolainen + * + * ALSA code based on ES18xx driver by Christian Fischbach & Abramo Bagnara + * and also by OPL3-SA2 driver by Jaroslav Kysela + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/isa.h> +#include <linux/pnp.h> +#include <linux/isapnp.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> +#include "waveartist.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("Driver for Rockwell WaveArtist RWA010 sound cards"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#ifdef CONFIG_PNP +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +#endif +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x250-0x3f0 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x388-0x3f0 */ +static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x300-0x3f0 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,10,11 */ +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */ +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for WaveArtist soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for WaveArtist soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable WaveArtist soundcard."); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for WaveArtist driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for WaveArtist driver."); +module_param_array(midi_port, long, NULL, 0444); +MODULE_PARM_DESC(midi_port, "MIDI port # for WaveArtist driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for WaveArtist driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 # for WaveArtist driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 # for WaveArtist driver."); + +#ifdef CONFIG_PNP +static int isa_registered; +static int pnp_registered; +#endif + +#define PFX "waveartist: " + +#ifdef CONFIG_PNP +#define WA_DEVICE(pnpid) { \ + .id = pnpid, \ + .devs = { {"RSS5000"}, {"RSS5001"}, {"RSS5002"} } \ +} +/* + * RSS5000 = WaveArtist, RSS5001 = SB, RSS5002 = MPU-401, RSS5003 = IDE, + * RSS5004 = gameport, RSS5005 = modem, RSS5006 = 3D + */ + +static struct pnp_card_device_id snd_waveartist_pnpids[] = { + WA_DEVICE("RSS5000"), /* 16-bit decode */ + WA_DEVICE("RSS5100"), /* 16-bit decode + modem */ + WA_DEVICE("RSS5200"), /* 10-bit decode + modem */ + WA_DEVICE("RSS5300"), /* 10-bit decode */ + WA_DEVICE("RSS5400"), /* 16-bit decode + IDE */ + WA_DEVICE("RSS5500"), /* 16-bit decode + modem + IDE */ + WA_DEVICE("RSS5600"), /* 10-bit decode + IDE */ + WA_DEVICE("RSS5700"), /* 10-bit decode + modem + IDE */ + WA_DEVICE("RSS5800"), /* 10-bit decode + modem + 3D */ + WA_DEVICE("RSS5900"), /* 10-bit decode + 3D */ + WA_DEVICE("RSS5A00"), /* 16-bit decode + modem + 3D */ + WA_DEVICE("RSS5B00"), /* 16-bit decode + 3D */ + { .id = "" } /* end */ +}; +MODULE_DEVICE_TABLE(pnp_card, snd_waveartist_pnpids); +#endif /* CONFIG_PNP */ + +struct snd_waveartist { +#ifdef CONFIG_PNP + struct pnp_dev *wa; /* WaveArtist device */ + struct pnp_dev *sb; /* SB emulation device */ + struct pnp_dev *mpu; /* MPU-401 device */ +#endif + unsigned long port; /* base port */ + struct resource *res_port; /* base port resource */ + int irq; + int dma_playback; + int dma_capture; + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + spinlock_t reg_lock; + struct snd_hwdep *synth; + struct snd_rawmidi *rmidi; + + u16 image[20]; /* mixer registers image */ +}; + +static inline void wa_outb(struct snd_waveartist *chip, u8 reg, u8 val) +{ + outb(val, chip->port + reg); +} + +static inline u8 wa_inb(struct snd_waveartist *chip, u8 reg) +{ + return inb(chip->port + reg); +} + +static inline void wa_outw(struct snd_waveartist *chip, u8 reg, u16 val) +{ + outw(val, chip->port + reg); +} + +static inline u16 wa_inw(struct snd_waveartist *chip, u8 reg) +{ + return inw(chip->port + reg); +} + +static inline void waveartist_set_ctlr(struct snd_waveartist *chip, u8 clear, + u8 set) +{ + clear = ~clear & wa_inb(chip, CTLR); + wa_outb(chip, CTLR, clear | set); +} + +/* acknowledge IRQ */ +static inline void waveartist_iack(struct snd_waveartist *chip) +{ + u8 old_ctlr = wa_inb(chip, CTLR) & ~IRQ_ACK; + + wa_outb(chip, CTLR, old_ctlr | IRQ_ACK); + wa_outb(chip, CTLR, old_ctlr); +} + +static int waveartist_reset(struct snd_waveartist *chip) +{ + unsigned int timeout, res = -1; + + waveartist_set_ctlr(chip, -1, RESET); + msleep(200); + waveartist_set_ctlr(chip, RESET, 0); + + timeout = 500; + do { + mdelay(2); + + if (wa_inb(chip, STATR) & CMD_RF) { + res = wa_inw(chip, CMDR); + if (res == 0x55aa) + break; + } + } while (--timeout); + + if (timeout == 0) { + dev_warn(chip->card->dev, "WaveArtist: reset timeout (res=0x%x)\n", + res); + return 1; + } + + return 0; +} + +/* Helper function to send and receive words + * from WaveArtist. It handles all the handshaking + * and can send or receive multiple words. + */ +static int waveartist_cmd(struct snd_waveartist *chip, + int nr_cmd, u16 *cmd, + int nr_resp, u16 *resp) +{ + unsigned long flags; + unsigned int timed_out = 0, i; + + spin_lock_irqsave(&chip->reg_lock, flags); + /* + * The chip can hang if we access the STATR register too quickly + * after a write. Do a dummy read to slow down. + */ + wa_inb(chip, CTLR); + + if (wa_inb(chip, STATR) & CMD_RF) { + /* flush the port */ + wa_inw(chip, CMDR); + udelay(10); + } + + for (i = 0; !timed_out && i < nr_cmd; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_WE) + break; + + if (!count) + timed_out = 1; + else + wa_outw(chip, CMDR, cmd[i]); + /* Another dummy read */ + wa_inb(chip, CTLR); + } + + for (i = 0; !timed_out && i < nr_resp; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_RF) + break; + + if (!count) + timed_out = 1; + else + resp[i] = wa_inw(chip, CMDR); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + + return timed_out ? 1 : 0; +} + +/* Send one command word */ +static inline int waveartist_cmd1(struct snd_waveartist *chip, u16 cmd) +{ + return waveartist_cmd(chip, 1, &cmd, 0, NULL); +} + +/* Send one command, receive one word */ +static inline u16 waveartist_cmd1_r(struct snd_waveartist *chip, u16 cmd) +{ + u16 ret; + + waveartist_cmd(chip, 1, &cmd, 1, &ret); + + return ret; +} + +/* Send a double command, receive one word (and throw it away) */ +static inline int waveartist_cmd2(struct snd_waveartist *chip, u16 cmd, u16 arg) +{ + u16 vals[2] = { cmd, arg }; + + return waveartist_cmd(chip, 2, vals, 1, vals); +} + +/* Send a triple command */ +static inline int waveartist_cmd3(struct snd_waveartist *chip, u16 cmd, + u16 arg1, u16 arg2) +{ + u16 vals[3] = { cmd, arg1, arg2 }; + + return waveartist_cmd(chip, 3, vals, 0, NULL); +} + +static u16 waveartist_getrev(struct snd_waveartist *chip) +{ + u16 temp[2]; + u16 cmd = WACMD_GETREV; + + waveartist_cmd(chip, 1, &cmd, 2, temp); + + return temp[0]; +} + +static irqreturn_t snd_waveartist_interrupt(int irq, void *dev_id) +{ + u8 status, irqstatus; + struct snd_card *card = dev_id; + struct snd_waveartist *chip; + + if (card == NULL) + return IRQ_NONE; + + chip = card->private_data; + + irqstatus = wa_inb(chip, IRQSTAT); + status = wa_inb(chip, STATR); + + if (status & IRQ_REQ) /* clear interrupt */ + waveartist_iack(chip); + + if (irqstatus & IRQ_PCM) { /* PCM buffer done */ + if ((status & DMA1) && chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + if ((status & DMA0) && chip->capture_substream) + snd_pcm_period_elapsed(chip->capture_substream); + if (!(status & (DMA0 | DMA1))) + dev_warn(chip->card->dev, "Unknown PCM interrupt\n"); + } + + if (irqstatus & IRQ_SB) /* we do not use SB mode */ + dev_warn(chip->card->dev, "Unexpected SB interrupt\n"); + + if ((irqstatus & IRQ_MPU) && chip->rmidi) + snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); + + return IRQ_HANDLED; +} + +#ifdef CONFIG_PM +static int snd_waveartist_suspend(struct snd_card *card, pm_message_t state) +{ + struct snd_waveartist *chip = card->private_data; + int i; + + if (!card) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + + /* save mixer registers */ + for (i = 0; i < ARRAY_SIZE(chip->image); i++) + chip->image[i] = waveartist_cmd1_r(chip, + WACMD_GET_LEVEL | i << 8); + + return 0; +} + +static int snd_waveartist_resume(struct snd_card *card) +{ + struct snd_waveartist *chip; + int i; + + if (!card) + return 0; + + chip = card->private_data; + + /* restore mixer registers */ + for (i = 0; i < 10; i += 2) + waveartist_cmd3(chip, WACMD_SET_MIXER, + chip->image[i], chip->image[i + 1]); + for (i = 10; i < ARRAY_SIZE(chip->image); i += 2) + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((i - 10) / 2) << 8, + chip->image[i], chip->image[i + 1]); + + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM */ + +#ifdef CONFIG_PNP +static void snd_waveartist_set_irq(struct pnp_dev *pdev, int irq) +{ + if (!pdev->active) + return; + isapnp_cfg_begin(isapnp_card_number(pdev), isapnp_csn_number(pdev)); + isapnp_write_byte(0x70, irq); /* ISAPNP_CFG_IRQ */ + isapnp_cfg_end(); +} + +static int snd_waveartist_pnp(int dev, struct snd_waveartist *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *id) +{ + chip->wa = pnp_request_card_device(card, id->devs[0].id, NULL); + if (!chip->wa) + return -EBUSY; + + chip->sb = pnp_request_card_device(card, id->devs[1].id, NULL); + if (!chip->sb) + return -EBUSY; + + chip->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (!chip->mpu) + return -EBUSY; + + if (pnp_activate_dev(chip->wa) < 0) { + dev_err(chip->card->dev, "WA PnP configure failure\n"); + return -EBUSY; + } + if (pnp_activate_dev(chip->sb) < 0) { + dev_err(chip->card->dev, "SB PnP configure failure\n"); + return -EBUSY; + } + port[dev] = pnp_port_start(chip->wa, 0); + dma2[dev] = pnp_dma(chip->wa, 0); + fm_port[dev] = pnp_port_start(chip->sb, 1); + dma1[dev] = pnp_dma(chip->sb, 0); + irq[dev] = pnp_irq(chip->sb, 0); + + /* + * The card uses only one IRQ (listed in the resources of SB device) + * which needs to be shared by WaveArtist and MPU-401 devices. They + * don't have an IRQ resource so it must be forced. + */ + snd_waveartist_set_irq(chip->wa, irq[dev]); + + /* allocate MPU-401 resources */ + if (pnp_activate_dev(chip->mpu) < 0) + dev_err(chip->card->dev, "MPU-401 PnP configure failure: will be disabled\n"); + else { + midi_port[dev] = pnp_port_start(chip->mpu, 0); + snd_waveartist_set_irq(chip->mpu, irq[dev]); + } + + dev_dbg(chip->card->dev, "PnP WaveArtist: port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", + port[dev], fm_port[dev], midi_port[dev]); + dev_dbg(chip->card->dev, "PnP WaveArtist: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); + + return 0; +} +#endif /* CONFIG_PNP */ + +static int snd_waveartist_playback_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static enum wa_format waveartist_format(snd_pcm_format_t format) +{ + if (snd_pcm_format_width(format) == 16) + return WA_FMT_S16; + if (snd_pcm_format_unsigned(format)) + return WA_FMT_U8; + else + return WA_FMT_S8; +} + +static u16 waveartist_rate(struct snd_pcm_runtime *runtime) +{ + return (runtime->rate << 16) / 44100; +} + +static int snd_waveartist_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting playback rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting playback %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_OUTPUTDMA, + chip->dma_playback > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting playback data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_OUTPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting playback format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting playback count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_playback, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_OUTPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_OUTPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_OUTPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_OUTPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_waveartist_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_INPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting capture rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_INPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting capture %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_INPUTDMA, + chip->dma_capture > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting capture data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_INPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting capture format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_INPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting capture count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_capture, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_INPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_INPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_INPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_INPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_waveartist_playback_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_playback, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static snd_pcm_uframes_t snd_waveartist_capture_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_capture, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static struct snd_pcm_hardware snd_waveartist_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_waveartist_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_waveartist_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = substream; + substream->runtime->hw = snd_waveartist_playback; + + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = substream; + substream->runtime->hw = snd_waveartist_capture; + + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_waveartist_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static struct snd_pcm_ops snd_waveartist_playback_ops = { + .open = snd_waveartist_playback_open, + .close = snd_waveartist_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_playback_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_playback_prepare, + .trigger = snd_waveartist_playback_trigger, + .pointer = snd_waveartist_playback_pointer, +}; + +static struct snd_pcm_ops snd_waveartist_capture_ops = { + .open = snd_waveartist_capture_open, + .close = snd_waveartist_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_capture_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_capture_prepare, + .trigger = snd_waveartist_capture_trigger, + .pointer = snd_waveartist_capture_pointer, +}; + +static int snd_waveartist_pcm(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "WaveArtist", 0, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_waveartist_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_waveartist_capture_ops); + + /* global setup */ + pcm->private_data = chip; + pcm->info_flags = 0; + if (chip->dma_playback == chip->dma_capture) + pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(pcm->name, card->shortname); + chip->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), 64 * 1024, + chip->dma_playback > 3 || chip->dma_capture > 3 ? + 128 * 1024 : 64 * 1024); + + return 0; +} + +static void snd_waveartist_free(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + + release_and_free_resource(chip->res_port); + if (chip->irq >= 0) + free_irq(chip->irq, card); + if (chip->dma_playback >= 0) { + disable_dma(chip->dma_playback); + free_dma(chip->dma_playback); + } + if (chip->dma_capture >= 0 && chip->dma_capture != chip->dma_playback) { + disable_dma(chip->dma_capture); + free_dma(chip->dma_capture); + } +} + +static int snd_waveartist_dev_free(struct snd_device *device) +{ + snd_waveartist_free(device->card); + + return 0; +} + +static int snd_waveartist_init(struct snd_waveartist *chip) +{ + if (waveartist_reset(chip)) + return -ENODEV; + dev_dbg(chip->card->dev, "chip rev. 0x%x\n", waveartist_getrev(chip)); + + waveartist_cmd1(chip, WACMD_RST_MIXER); /* reset mixer */ + waveartist_iack(chip); /* clear any pending interrupt */ + waveartist_set_ctlr(chip, 0, DMA1_IE | DMA0_IE); /* enable DMA IRQs */ + waveartist_iack(chip); /* clear any pending interrupt */ + + return 0; +} + +static int snd_waveartist_new_device(struct snd_card *card, + unsigned long port, + unsigned long mpu_port, + unsigned long fm_port, + int irq, int dma1, int dma2) +{ + struct snd_waveartist *chip = card->private_data; + static struct snd_device_ops ops = { + .dev_free = snd_waveartist_dev_free, + }; + int err; + + spin_lock_init(&chip->reg_lock); + chip->card = card; + chip->port = port; + chip->irq = -1; + chip->dma_playback = -1; + chip->dma_capture = -1; + + chip->res_port = request_region(port, 16, "WaveArtist"); + if (!chip->res_port) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab ports 0x%lx-0x%lx\n", + port, port + 16 - 1); + return -EBUSY; + } + + if (snd_waveartist_init(chip) < 0) { + snd_waveartist_free(card); + return -ENODEV; + } + + if (request_irq(irq, snd_waveartist_interrupt, 0, "WaveArtist", card)) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab IRQ %d\n", irq); + return -EBUSY; + } + chip->irq = irq; + + if (dma1 == dma2 || dma1 == SNDRV_AUTO_DMA || dma2 == SNDRV_AUTO_DMA) { + /* we have only one DMA channel */ + if (dma1 == SNDRV_AUTO_DMA) + dma1 = dma2; + if (request_dma(dma1, "WaveArtist")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = chip->dma_capture = dma1; + } else { + /* + * 2 channels: use 16-bit for playback and 8-bit for capture as + * full-duplex works better this way. However, the chip seems to + * have some band-width limit so full-duplex at 44kHz/16-bit/2ch + * is not possible - some frames are dropped during capture, + * resulting in too-fast recording. If capture is done using + * lower rate or 8-bit or mono, everything is fine. + */ + if (dma2 > 3) + swap(dma1, dma2); + + if (request_dma(dma1, "WaveArtist playback")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab playback DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = dma1; + + if (dma2 != dma1 && request_dma(dma2, "WaveArtist capture")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab capture DMA %d\n", + dma2); + return -EBUSY; + } + chip->dma_capture = dma2; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_waveartist_free(card); + return err; + } + + return 0; +} + +static int snd_waveartist_card_new(struct device *pdev, int dev, + struct snd_card **cardp) +{ + return snd_card_new(pdev, index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_waveartist), cardp); +} + +static int snd_waveartist_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_waveartist_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val; + + val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + ucontrol->value.integer.value[0] = (val >> shift) & mask; + + return 0; +} + +static int snd_waveartist_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val, old_val, new_val; + + val = (ucontrol->value.integer.value[0] & mask); + mask <<= shift; + val <<= shift; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + new_val = (old_val & ~mask) | (val & mask); + + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +static int snd_waveartist_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask ? mask : 0x7fff; + + return 0; +} + +static int snd_waveartist_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 left, right; + + if (mask == 0) + mask = 0x7fff; + + left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + right = left; + + ucontrol->value.integer.value[0] = (left >> shift_left) & mask; + ucontrol->value.integer.value[1] = (right >> shift_right) & mask; + + return 0; +} + +static int snd_waveartist_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 val1, val2, mask1, mask2; + u16 old_left, old_right, new_left, new_right; + + if (mask == 0) + mask = 0x7fff; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + val1 <<= shift_left; + val2 <<= shift_right; + mask1 = mask << shift_left; + mask2 = mask << shift_right; + + old_left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + old_right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + old_right = old_left; + + new_left = (old_left & ~mask1) | (val1 & mask1); + new_right = (old_right & ~mask2) | (val2 & mask2); + + if (new_left == old_left && new_right == old_right) + return 0; + + if (left_reg < 10) + /* new_* already contain reg. num. (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_left, new_right); + else + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((left_reg - 10) / 2) << 8, + new_left, new_right); + + return 1; +} + +static int snd_waveartist_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const mux_texts[] = { + "None", "Mix", "Line", "Phone", "CD", "Mic" + }; + + return snd_ctl_enum_info(uinfo, 2, ARRAY_SIZE(mux_texts), mux_texts); +} + +static int snd_waveartist_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int mux = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + + ucontrol->value.enumerated.item[0] = mux & 0x07; + ucontrol->value.enumerated.item[1] = (mux >> 3) & 0x07; + + return 0; +} + +static int snd_waveartist_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + u16 old_val, new_val; + u16 val1 = ucontrol->value.enumerated.item[0]; + u16 val2 = ucontrol->value.enumerated.item[1]; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + new_val = (old_val & ~0x3f) | (val1 & 0x07) | ((val2 & 0x07) << 3); + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +#define WAVEARTIST_SINGLE(xname, reg, shift, mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_single, \ + .get = snd_waveartist_get_single, .put = snd_waveartist_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) \ +} + +#define WAVEARTIST_DOUBLE(xname, left_reg, right_reg, shift_left, shift_right, \ + mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_double, \ + .get = snd_waveartist_get_double, .put = snd_waveartist_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 20) | (mask << 24) \ +} + +static struct snd_kcontrol_new snd_waveartist_controls[] = { +WAVEARTIST_DOUBLE("Master Playback Volume", 2, 6, 1, 1, 0x07), +WAVEARTIST_DOUBLE("Master Playback Switch", 0, 4, 0, 0, 1), +WAVEARTIST_DOUBLE("PCM Playback Volume", 10, 11, 0, 0, 0x00),/* 0x00 = 0x7fff */ +WAVEARTIST_DOUBLE("FM Playback Volume", 12, 13, 0, 0, 0x00),/* 0x00 = 0x7fff */ +/* WAVEARTIST_DOUBLE("Wavetable? Playback Volume", 14, 15, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 16, 17, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 18, 19, 0, 0, 0x00), */ +WAVEARTIST_DOUBLE("Digital Playback Switch", 3, 7, 10, 10, 1), /* PCM + FM */ +WAVEARTIST_DOUBLE("CD Playback Volume", 0, 4, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("CD Playback Switch", 3, 7, 6, 6, 1), +WAVEARTIST_DOUBLE("Line Playback Volume", 0, 4, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Line Playback Switch", 3, 7, 4, 4, 1), +WAVEARTIST_DOUBLE("Phone Playback Volume", 1, 5, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Phone Playback Switch", 3, 7, 5, 5, 1), +WAVEARTIST_SINGLE("Mono Playback Volume", 8, 5, 0x1f), +WAVEARTIST_DOUBLE("Mono Playback Switch", 3, 7, 9, 9, 1), +WAVEARTIST_DOUBLE("Mic Playback Volume", 2, 6, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Mic 2 Playback Volume", 1, 5, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("Mic Playback Switch", 3, 7, 7, 7, 1), +WAVEARTIST_DOUBLE("Mic 2 Playback Switch", 3, 7, 8, 8, 1), +WAVEARTIST_DOUBLE("Mic Gain", 2, 6, 4, 4, 0x03), +WAVEARTIST_DOUBLE("Capture Volume", 3, 7, 0, 0, 0x0f), +WAVEARTIST_SINGLE("Mono Output Playback Switch", 1, 0, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_waveartist_info_mux, + .get = snd_waveartist_get_mux, + .put = snd_waveartist_put_mux, +} +}; + +static int snd_waveartist_mixer(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + int err; + unsigned int idx; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_waveartist_controls); idx++) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1(&snd_waveartist_controls[idx], chip); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_waveartist_probe(struct snd_card *card, int dev) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_opl3 *opl3; + int err; + + err = snd_waveartist_new_device(card, + port[dev], midi_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) + return err; + + strcpy(card->driver, "WaveArtist"); + strcpy(card->shortname, "Rockwell WaveArtist RWA010"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d", + card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]); + + err = snd_waveartist_pcm(card); + if (err < 0) + return err; + err = snd_waveartist_mixer(card); + if (err < 0) + return err; + + if (fm_port[dev] >= 0x388 && fm_port[dev] < 0x3f0) { + err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3); + if (err < 0) + return err; + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + return err; + err = snd_opl3_hwdep_new(opl3, 0, 1, &chip->synth); + if (err < 0) + return err; + } + if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x3f0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_WAVEARTIST, + midi_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); + if (err < 0) + return err; + } + + return snd_card_register(card); +} + +static int snd_waveartist_isa_match(struct device *pdev, + unsigned int dev) +{ + if (!enable[dev]) + return 0; +#ifdef CONFIG_PNP + if (isapnp[dev]) + return 0; +#endif + if (port[dev] == SNDRV_AUTO_PORT) { + dev_err(pdev, "specify port\n"); + return 0; + } + if (irq[dev] == SNDRV_AUTO_IRQ) { + dev_err(pdev, "specify irq\n"); + return 0; + } + if (dma1[dev] == SNDRV_AUTO_DMA) { + dev_err(pdev, "specify dma1\n"); + return 0; + } + + return 1; +} + +static int snd_waveartist_isa_probe(struct device *pdev, + unsigned int dev) +{ + struct snd_card *card; + int err; + + err = snd_waveartist_card_new(pdev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; +} + +static int snd_waveartist_isa_remove(struct device *devptr, + unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_waveartist_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_waveartist_suspend(dev_get_drvdata(dev), state); +} + +static int snd_waveartist_isa_resume(struct device *dev, unsigned int n) +{ + return snd_waveartist_resume(dev_get_drvdata(dev)); +} +#endif + +static struct isa_driver waveartist_isa_driver = { + .match = snd_waveartist_isa_match, + .probe = snd_waveartist_isa_probe, + .remove = snd_waveartist_isa_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_isa_suspend, + .resume = snd_waveartist_isa_resume, +#endif + .driver = { .name = "waveartist" }, +}; + +#ifdef CONFIG_PNP +static int snd_waveartist_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int dev; + int err; + struct snd_card *card; + + for (; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + if (dev >= SNDRV_CARDS) + return -ENODEV; + + err = snd_waveartist_card_new(&pcard->card->dev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_pnp(dev, card->private_data, pcard, pid); + if (err < 0) { + dev_err(&pcard->card->dev, "PnP detection failed\n"); + snd_card_free(card); + return err; + } + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + dev++; + + return 0; +} + +static void snd_waveartist_pnp_remove(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* disable forced IRQs */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +#ifdef CONFIG_PM +static int snd_waveartist_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + snd_waveartist_suspend(card, state); + + /* disable forced IRQs to prevent opps */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + return 0; +} +static int snd_waveartist_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* re-enable forced IRQs */ + snd_waveartist_set_irq(chip->wa, chip->irq); + snd_waveartist_set_irq(chip->mpu, chip->irq); + + return snd_waveartist_resume(pnp_get_card_drvdata(pcard)); +} +#endif + +static struct pnp_card_driver waveartist_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "waveartist", + .id_table = snd_waveartist_pnpids, + .probe = snd_waveartist_pnp_detect, + .remove = snd_waveartist_pnp_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_pnp_suspend, + .resume = snd_waveartist_pnp_resume, +#endif +}; +#endif /* CONFIG_PNP */ + +static int __init alsa_card_waveartist_init(void) +{ + int err; + + err = isa_register_driver(&waveartist_isa_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&waveartist_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit alsa_card_waveartist_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&waveartist_pnpc_driver); + + if (isa_registered) +#endif + isa_unregister_driver(&waveartist_isa_driver); +} + +module_init(alsa_card_waveartist_init) +module_exit(alsa_card_waveartist_exit) diff --git a/sound/isa/waveartist.h b/sound/isa/waveartist.h new file mode 100644 index 0000000..72254cf --- /dev/null +++ b/sound/isa/waveartist.h @@ -0,0 +1,74 @@ +/* + * Rockwell WaveArtist RWA010 chip register definitions + * + * Based on OSS WaveArtist driver by Hannu Savolainen + */ + +/* registers */ +#define CMDR 0 /* command (16-bit) */ +#define DATR 2 /* data (for PIO?) (16-bit) */ +#define CTLR 4 /* control */ +#define STATR 5 /* status */ +#define EXPCR1 6 /* expansion control 1 */ +#define EXPCR2 7 /* expansion control 2 */ +#define EXPDAT1 8 /* expansion data 1 (16-bit) */ +#define EXPDAT2 10 /* expansion data 2 (16-bit) */ +#define IRQSTAT 12 /* IRQ status */ + +/* STATR register bit definitions */ +#define CMD_WE BIT(7) /* CMDR write empty (ready for write) */ +#define CMD_RF BIT(6) /* CMDR read full (ready for read) */ +#define DAT_WE BIT(5) /* DATR write empty (ready for write) */ +#define DAT_RF BIT(4) /* DATR read full (ready for read) */ +#define IRQ_REQ BIT(3) /* IRQ was requested */ +#define DMA1 BIT(2) /* DMA1 IRQ was requested */ +#define DMA0 BIT(1) /* DMA0 IRQ was requested */ + +/* CTLR register bit definitions */ +#define CMD_WEIE BIT(7) /* CMDR write empty interrupt enable */ +#define CMD_RFIE BIT(6) /* CMDR read full interrupt enable */ +#define DAT_WEIE BIT(5) /* DATR write empty interrupt enable */ +#define DAT_RFIE BIT(4) /* DATR read full interrupt enable */ +#define RESET BIT(3) /* chip reset */ +#define DMA1_IE BIT(2) /* DMA1 interrupt enable */ +#define DMA0_IE BIT(1) /* DMA0 interrupt enable */ +#define IRQ_ACK BIT(0) /* IRQ acknowlege */ + +/* IRQSTAT register bit definitions */ +#define IRQ_MPU BIT(2) /* MPU-401 */ +#define IRQ_SB BIT(1) /* SB emulation */ +#define IRQ_PCM BIT(0) /* native PCM */ + +/* commands */ +#define WACMD_GETREV 0x00 + +#define WACMD_INPUTFORMAT 0x10 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_INPUTCHANNELS 0x11 /* 1=mono, 2=Stereo */ +#define WACMD_INPUTSPEED 0x12 /* sampling rate */ +#define WACMD_INPUTDMA 0x13 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_INPUTSIZE 0x14 /* samples to interrupt */ +#define WACMD_INPUTSTART 0x15 /* start ADC */ +#define WACMD_INPUTPAUSE 0x16 /* pause ADC */ +#define WACMD_INPUTSTOP 0x17 /* stop ADC */ +#define WACMD_INPUTRESUME 0x18 /* resume ADC */ +#define WACMD_INPUTPIO 0x19 /* PIO ADC */ + +#define WACMD_OUTPUTFORMAT 0x20 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_OUTPUTCHANNELS 0x21 /* 1=mono, 2=stereo */ +#define WACMD_OUTPUTSPEED 0x22 /* sampling rate */ +#define WACMD_OUTPUTDMA 0x23 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_OUTPUTSIZE 0x24 /* samples to interrupt */ +#define WACMD_OUTPUTSTART 0x25 /* start DAC */ +#define WACMD_OUTPUTPAUSE 0x26 /* pause DAC */ +#define WACMD_OUTPUTSTOP 0x27 /* stop DAC */ +#define WACMD_OUTPUTRESUME 0x28 /* resume DAC */ +#define WACMD_OUTPUTPIO 0x29 /* PIO DAC */ + +#define WACMD_GET_LEVEL 0x30 /* read mixer reg */ +#define WACMD_SET_LEVEL 0x31 /* set mixer regs (10..19) */ +#define WACMD_SET_MIXER 0x32 /* set mixer regs (0..9) */ +#define WACMD_RST_MIXER 0x33 /* mixer reset */ +#define WACMD_SET_MONO 0x34 /* set mono mode (|0x000=L, |0x100=R) */ + +enum wa_format { WA_FMT_S8 = 0, WA_FMT_S16, WA_FMT_U8 }; +enum wa_dma { WA_DMA_8BIT = 0, WA_DMA_16BIT, WA_DMA_PIO }; -- Ondrej Zary

1 1

[alsa-devel] PATCH: ALSA: hda: fix possible null dereference
by Markus Osterhoff 24 Aug '15

24 Aug '15

Upgrading on Gentoo hardened from 4.0.8 to 4.1.6 (with intermediate 4.1.4), I noticed a NULL pointer derefence during booting, and tackled it down to static int add_std_chmaps(struct hda_codec *codec) in file sound/pci/hda/hda_codec.c With commits bbbc7e85 and 751e2216, a for-loop was restated using list_for_each_entry(...); originally, a local pcm* was extracted from an array and checked against NULL, if not-NULL then passed along to snd_pcm_add_chmap_ctls(...). Now, pcm->pcm is passed to snd_pcm_add_chmap_ctls, but the NULL pointer check still uses the “upper level” pcm, not the “in-loop” pcm->pcm. Please have a look at the attached (rather small) suggestion to fix this problem. For a history of the function in question, have a look at lines 2430ff for git diff v4.0 v4.1 sound/pci/hda/hda_codec.c The two commits that touched this function are Commit bbbc7e8502c9 ("ALSA: hda - Allocate hda_pcm objects dynamically") Commit 751e2216899c ("ALSA: hda: fix possible null dereference") where the latter only fixes the order of two fixes, but not the pcm / pcm->pcm confusion. Using v4.1-descendant sources, boot got OOPS like this (two bug-hunting printk's added up front): > pcm: ffff88040cf83400 > pcm->pcm: (nil) > BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 > IP: [<ffffffffa07feab9>] snd_pcm_add_chmap_ctls+0x119/0x1f0 [snd_pcm] > PGD 0 > Oops: 0000 [#1] PREEMPT SMP NULL checking pcm->pcm instead of pcm fixes this problem. Greetings and thanks for your time, Markus Signed-off-by: Markus Osterhoff <linux-kernel(a)k-raum.org> --- --- sound/pci/hda/hda_codec.c 2015-08-23 20:02:57.281782648 +0200 +++ sound/pci/hda/hda_codec.c.orig 2015-08-23 20:02:49.723783063 +0200 @@ -3172,7 +3172,7 @@ static int add_std_chmaps(struct hda_cod struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (!pcm->pcm || pcm->own_chmap || + if (!pcm || pcm->own_chmap || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps;

2 5

[alsa-devel] [PATCH v3] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 24 Aug '15

24 Aug '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. No verification is made that the playback and capture codec setups are compatible in terms of number of TDM slots (where applicable). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V3: Updated commit message regarding the fact that the patch does not consider the slot setup in a TDM system with separate capture and playback codecs, after a comment from Pierre-Louis Bossart. Such a setup was not supported anyway before this patch, so let's handle that situation when and if it arises. V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: hdac: Fix size allocation for ext device allocation
by Vinod Koul 23 Aug '15

23 Aug '15

While creating hdac_ext_device, we used hdev for sizeof insteadof edev, which resulted in eventual crash of the system Fix the size here Fixes: a512f5611646 ('ALSA: hdac: add hdac extended device') Reported-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 8544e4fb1cd3..4449d1a99089 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -139,7 +139,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) char name[15]; int ret; - edev = kzalloc(sizeof(*hdev), GFP_KERNEL); + edev = kzalloc(sizeof(*edev), GFP_KERNEL); if (!edev) return -ENOMEM; hdev = &edev->hdac; -- 1.9.1

2 3

[alsa-devel] [PATCH 0/5] ALSA: hdac: More hdac core fixes and updates
by Vinod Koul 23 Aug '15

23 Aug '15

Hi Takashi, We had few more hdac core fixes & enhancements so senidng those. First one is fix for register pointers for spcap and link registers. This is required for SKL driver when we use link registers. rest add new APIs for powering down all links, getting stream from stream tag and direction, enabling SPIB. Last one remove keys as we dont need them for host streams Jeeja KP (5): ALSA: hdac: Fix to read the correct offset of spcap/link register ALSA: hdac: Add snd_hdac_ext_bus_link_power_down_all() ALSA: hdac: Add snd_hdac_get_hdac_stream() ALSA: hdac: Add support to enable SPIB for hdac ext stream ALSA: hdac: Remove the usage of key for host stream include/sound/hda_register.h | 4 +++ include/sound/hdaudio.h | 2 ++ include/sound/hdaudio_ext.h | 11 +++++++ sound/hda/ext/hdac_ext_controller.c | 30 +++++++++++++++--- sound/hda/ext/hdac_ext_stream.c | 62 ++++++++++++++++++++++++++++++++----- sound/hda/hdac_stream.c | 19 ++++++++++++ 6 files changed, 116 insertions(+), 12 deletions(-) Thanks -- ~Vinod

3 10

[alsa-devel] [PATCH v2 0/3] ASoC: topology: update kcontrol ops binding & support TLV bytes controls
by mengdong.lin＠intel.com 22 Aug '15

22 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> History: v2: - let vendor specific kcontrol handlers override standard ones. - Merge handler binding for TLV bytes controls into soc_tplg_kcontrol_bind_io() Mengdong Lin (3): ASoC: topology: Bind vendor specific kcontrol handlers before standard ones ASoC: topology: Reduce arguments of soc_tplg_kcontrol_bind_io() ASoC: topology: Add support for TLV bytes controls include/sound/soc-topology.h | 13 +++++- sound/soc/soc-topology.c | 101 ++++++++++++++++++++++++++++--------------- 2 files changed, 79 insertions(+), 35 deletions(-) -- 1.9.1

2 6

[alsa-devel] [patch] ALSA: hdsp: silence a sprinft() overflow warning
by Dan Carpenter 22 Aug '15

22 Aug '15

card->shortname is a 32 char string so the sprintf() can theoretically overflow. snd_rawmidi_new() can accept strings up to 64 bytes long. I have made the temporay buf[] array 40 bytes long and changed the sprintf() to snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..c1fe1d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1526,7 +1526,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) { - char buf[32]; + char buf[40]; hdsp->midi[id].id = id; hdsp->midi[id].rmidi = NULL; @@ -1537,7 +1537,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i hdsp->midi[id].pending = 0; spin_lock_init (&hdsp->midi[id].lock); - sprintf (buf, "%s MIDI %d", card->shortname, id+1); + snprintf(buf, sizeof(buf), "%s MIDI %d", card->shortname, id + 1); if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1;

2 4

[alsa-devel] [PATCH v2] ALSA: ac97: Switch to dev_pm_ops
by Lars-Peter Clausen 21 Aug '15

21 Aug '15

Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single AC'97 driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. While we are at it also drop the ifdefs around the suspend/resume functions to increase compile test coverage. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch touches code in both ALSA and the input subsystem. Ideally this patch will be merged via the ALSA tree. Given that the wm97xx touchscreen driver is not seeing too many changes these days the risk of conflicts should be low. Changes since v1: * Drop ifdefs around suspend/resum functions --- drivers/input/touchscreen/wm97xx-core.c | 13 ++++--------- sound/ac97_bus.c | 26 -------------------------- 2 files changed, 4 insertions(+), 35 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index b1ae779..1534e9b 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -732,8 +732,7 @@ static int wm97xx_remove(struct device *dev) return 0; } -#ifdef CONFIG_PM -static int wm97xx_suspend(struct device *dev, pm_message_t state) +static int __maybe_unused wm97xx_suspend(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); u16 reg; @@ -765,7 +764,7 @@ static int wm97xx_suspend(struct device *dev, pm_message_t state) return 0; } -static int wm97xx_resume(struct device *dev) +static int __maybe_unused wm97xx_resume(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); @@ -799,10 +798,7 @@ static int wm97xx_resume(struct device *dev) return 0; } -#else -#define wm97xx_suspend NULL -#define wm97xx_resume NULL -#endif +static SIMPLE_DEV_PM_OPS(wm97xx_pm_ops, wm97xx_suspend, wm97xx_resume); /* * Machine specific operations @@ -836,8 +832,7 @@ static struct device_driver wm97xx_driver = { .owner = THIS_MODULE, .probe = wm97xx_probe, .remove = wm97xx_remove, - .suspend = wm97xx_suspend, - .resume = wm97xx_resume, + .pm = &wm97xx_pm_ops, }; static int __init wm97xx_init(void) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe..57a6dfc 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -27,35 +27,9 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return 1; } -#ifdef CONFIG_PM -static int ac97_bus_suspend(struct device *dev, pm_message_t state) -{ - int ret = 0; - - if (dev->driver && dev->driver->suspend) - ret = dev->driver->suspend(dev, state); - - return ret; -} - -static int ac97_bus_resume(struct device *dev) -{ - int ret = 0; - - if (dev->driver && dev->driver->resume) - ret = dev->driver->resume(dev); - - return ret; -} -#endif /* CONFIG_PM */ - struct bus_type ac97_bus_type = { .name = "ac97", .match = ac97_bus_match, -#ifdef CONFIG_PM - .suspend = ac97_bus_suspend, - .resume = ac97_bus_resume, -#endif /* CONFIG_PM */ }; static int __init ac97_bus_init(void) -- 2.1.4

4 6

[alsa-devel] [PATCH] ASoC: Set missing card owner field
by Axel Lin 21 Aug '15

21 Aug '15

Set the card owner field to prevent the module from being removed from underneath its users. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/blackfin/bfin-eval-adau1x61.c | 1 + sound/soc/intel/boards/byt-max98090.c | 1 + sound/soc/intel/boards/byt-rt5640.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 2 ++ sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 1 + sound/soc/rockchip/rockchip_max98090.c | 1 + sound/soc/rockchip/rockchip_rt5645.c | 1 + sound/soc/samsung/arndale_rt5631.c | 1 + sound/soc/samsung/snow.c | 1 + 13 files changed, 14 insertions(+) diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c index 4229f76..fddfe00c 100644 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ b/sound/soc/blackfin/bfin-eval-adau1x61.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { static struct snd_soc_card bfin_eval_adau1x61 = { .name = "bfin-eval-adau1x61", + .owner = THIS_MODULE, .driver_name = "eval-adau1x61", .dai_link = &bfin_eval_adau1x61_dai, .num_links = 1, diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9..d9f81b8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -126,6 +126,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { static struct snd_soc_card byt_max98090_card = { .name = "byt-max98090", + .owner = THIS_MODULE, .dai_link = byt_max98090_dais, .num_links = ARRAY_SIZE(byt_max98090_dais), .dapm_widgets = byt_max98090_widgets, diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b9..de9788a 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -197,6 +197,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { static struct snd_soc_card byt_rt5640_card = { .name = "byt-rt5640", + .owner = THIS_MODULE, .dai_link = byt_rt5640_dais, .num_links = ARRAY_SIZE(byt_rt5640_dais), .dapm_widgets = byt_rt5640_widgets, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59..c445312 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -185,6 +185,7 @@ static struct snd_soc_dai_link byt_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_byt = { .name = "baytrailcraudio", + .owner = THIS_MODULE, .dai_link = byt_dailink, .num_links = ARRAY_SIZE(byt_dailink), .dapm_widgets = byt_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 7810fc0..49f4869 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -275,6 +275,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "chtmax98090", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .aux_dev = &cht_max98090_headset_dev, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..7be8461 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -305,6 +305,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, @@ -317,6 +318,7 @@ static struct snd_soc_card snd_soc_card_chtrt5645 = { static struct snd_soc_card snd_soc_card_chtrt5650 = { .name = "chtrt5650", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..23fe040 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -323,6 +323,7 @@ static int cht_resume_post(struct snd_soc_card *card) /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 39e5660..684e8a7 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -136,6 +136,7 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { static struct snd_soc_card mt8173_max98090_card = { .name = "mt8173-max98090", + .owner = THIS_MODULE, .dai_link = mt8173_max98090_dais, .num_links = ARRAY_SIZE(mt8173_max98090_dais), .controls = mt8173_max98090_controls, diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index b722145..86cf975 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -191,6 +191,7 @@ static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = { static struct snd_soc_card mt8173_rt5650_rt5676_card = { .name = "mtk-rt5650-rt5676", + .owner = THIS_MODULE, .dai_link = mt8173_rt5650_rt5676_dais, .num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais), .codec_conf = mt8173_rt5650_rt5676_codec_conf, diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index acace20..cc26f81 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -147,6 +147,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "ROCKCHIP-I2S", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .aux_dev = &rk_98090_headset_dev, diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 3c6bb1e..0940279 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link rk_dailink = { static struct snd_soc_card snd_soc_card_rk = { .name = "I2S-RT5650", + .owner = THIS_MODULE, .dai_link = &rk_dailink, .num_links = 1, .dapm_widgets = rk_dapm_widgets, diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 9e371eb..ee1fda9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -71,6 +71,7 @@ static struct snd_soc_dai_link arndale_rt5631_dai[] = { static struct snd_soc_card arndale_rt5631 = { .name = "Arndale RT5631", + .owner = THIS_MODULE, .dai_link = arndale_rt5631_dai, .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..07ce2cf 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -56,6 +56,7 @@ static int snow_late_probe(struct snd_soc_card *card) static struct snd_soc_card snow_snd = { .name = "Snow-I2S", + .owner = THIS_MODULE, .dai_link = snow_dai, .num_links = ARRAY_SIZE(snow_dai), -- 2.1.0

2 1

Re: [alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 21 Aug '15

21 Aug '15

On Fri, 21 Aug 2015, Pierre-Louis Bossart wrote: > On 8/21/15 2:11 AM, Ricard Wanderlof wrote: > > > > On Thu, 20 Aug 2015, Pierre-Louis Bossart wrote: > > > >>> /** > >>> * snd_soc_runtime_activate() - Increment active count for PCM runtime components > >>> * @rtd: ASoC PCM runtime that is activated > >>> @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) > >>> > >>> /* first calculate min/max only for CODECs in the DAI link */ > >>> for (i = 0; i < rtd->num_codecs; i++) { > >>> + > >>> + /* > >>> + * Skip CODECs which don't support the current stream type. > >>> + * Otherwise, since the rate, channel, and format values will > >>> + * zero in that case, we would have no usable settings left, > >>> + * causing the resulting setup to fail. > >>> + * At least one CODEC should match, otherwise we should have > >>> + * bailed out on a higher level, since there would be no > >>> + * CODEC to support the transfer direction in that case. > >>> + */ > >>> + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], > >>> + substream->stream)) > >> > >> Maybe I misunderstood but shouldn't there be some sort of verification > >> that the codecs can use the same number of slots between playback and > >> capture if they share the same LRCLK/FS? e.g. it's not uncommon to have > >> 4 mic capture and 2 ch playback. If the capture and playback is handled > >> by different chips you'd still need to maintain some level of consistency. > > > > You're probably right, although it may be that that sort of thing is > > handled at a higher level, e.g. the machine driver in such a setup would > > configure both codecs to use TDM before we even get to this function, I > > don't know. > > > > I think we'd have to take care of that situation if and when it arises. > > The setup I have here doesn't allow me to test it. > > ok, sounds fine. maybe you could add this to the comments or commit > message then so that this assumption is known. Ok, good point. Will do. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH] ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs
by Takashi Iwai 21 Aug '15

21 Aug '15

Intel Haswell (and later) codec refreshes the widgets tree to expose the whole widget nodes at probing. However, this refresh was missing for sysfs tree. This patch adds the missing piece, as we have now a proper API. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a97db5fc8a15..df982fc6a872 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2331,6 +2331,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (is_haswell_plus(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + snd_hdac_refresh_widget_sysfs(&codec->core); } /* For Valleyview/Cherryview, only the display codec is in the display -- 2.5.0

1 0

[alsa-devel] [patch] ALSA: hdsp: silence and underflow warning
by Dan Carpenter 21 Aug '15

21 Aug '15

I believe this probably cannot happen, as the code suggests. There would have to be an kcontrol->index.id which was zero, otherwise this would be prevented in snd_ctl_find_id(). But snd_BUG_ON() is just a WARN() or a no-op so static checkers complain that we keep on going with a negative offset. Let's just handle the error as well as printing a warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..468a95c 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2806,7 +2806,8 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); offset = ucontrol->id.index - 1; - snd_BUG_ON(offset < 0); + if (snd_BUG_ON(offset < 0)) + return -EINVAL; switch (hdsp->io_type) { case Digiface:

2 1

[alsa-devel] [PATCH 0/4] ALSA: hdac: add codec driver support
by Vinod Koul 21 Aug '15

21 Aug '15

This patch adds the core framework support for adding hdac extended devices and drivers. We introduce a minimal hdac extended device and driver and APIs for registration This series also ads extended device cleanup API and refresh sysfs tree API Vinod Koul (4): ALSA: hdac: Add API for removing hdac extended device ALSA: hdac: add hdac extended device ALSA: hdac: add extended device driver registration ALSA: hdac: add snd_hdac_refresh_widget_sysfs() include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 60 +++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_bus.c | 98 ++++++++++++++++++++++++++++++++++++++++++-- sound/hda/hdac_device.c | 28 +++++++++++++ 4 files changed, 183 insertions(+), 4 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH] usb: Add native DSD support for Gustard DAC-X20U
by Jurgen Kramer 21 Aug '15

21 Aug '15

This patch adds native DSD support for the Gustard DAC-X20U. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 754e689..00ebc0c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1268,6 +1268,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */ case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ if (fp->altsetting == 3) -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: hda - Shutdown CX20722 on reboot/free to avoid spurious noises
by David Henningsson 21 Aug '15

21 Aug '15

On shutdown/reboot of CX20722, first shut down all EAPDs, then shut down the afg node to D3. Failure to do so can lead to spurious noises from the internal speaker directly after reboot (and before the codec is reinitialized again, i e in BIOS setup or GRUB menus). BugLink: https://bugs.launchpad.net/bugs/1487345 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f788a91..ca03c40 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -200,12 +200,33 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -#define cx_auto_free snd_hda_gen_free +static void cx_auto_reboot_notify(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + + if (codec->core.vendor_id != 0x14f150f2) + return; + + /* Turn the CX20722 codec into D3 to avoid spurious noises + from the internal speaker during (and after) reboot */ + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); + + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} + +static void cx_auto_free(struct hda_codec *codec) +{ + cx_auto_reboot_notify(codec); + snd_hda_gen_free(codec); +} static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, + .reboot_notify = cx_auto_reboot_notify, .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM -- 1.9.1

2 3

[alsa-devel] [PATCH 0/3] Add ASoC support for AMD APUs [v2]
by Alex Deucher 21 Aug '15

21 Aug '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create an mfd cell which we can hang the i2s codec on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review There was a question about the code and the module licensing. IANAL, but this follows the same pattern that most if not all drm drivers use. There was also a comment about the patch changelog. I think this is more of a subsystem preference kind of thing. Generally in the drm we include the patch changelog and versions in the commit messages so we have a history of the patch if we need to reference it in the future. If you have a strong preference for the audio patch, we can drop that. Patch 1 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream2 Thanks, Alex Chunming Zhou (1): drm/amd: add ACP 2.x register headers Maruthi Bayyavarapu (1): drm/amd: add ACP driver support [v6] Maruthi Srinivas Bayyavarapu (1): ASoC: AMD: add AMD ASoC ACP-I2S driver [v4] drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 1133 ++++++++++ drivers/gpu/drm/amd/acp/acp_hw.h | 91 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 5 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 208 ++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 40 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + .../gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h | 437 ++++ .../drm/amd/include/asic_reg/acp/acp_2_1_enum.h | 1198 ++++++++++ .../drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h | 1568 +++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h | 609 ++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_enum.h | 1068 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h | 582 +++++ .../drm/amd/include/asic_reg/acp/acp_2_3_enum.h | 1079 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h | 2192 +++++++++++++++++++ include/linux/mfd/amd_acp.h | 211 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++ 27 files changed, 13494 insertions(+), 1 deletion(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.h create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c -- 1.8.3.1

2 10

[alsa-devel] [PATCH] ASoC: Samsung: Remove redundant arndale_audio_remove
by Vaishali Thakkar 21 Aug '15

21 Aug '15

There is no use of snd_soc_unregister_card in remove function as devm_snd_soc_register_card in probe function automatically handles it. So, remove use of snd_soc_unregister_card and with this change remove arndale_audio_remove as it is now redundant. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/samsung/arndale_rt5631.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 8bf2e2c..9e371eb 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -116,15 +116,6 @@ static int arndale_audio_probe(struct platform_device *pdev) return ret; } -static int arndale_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -139,7 +130,6 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, - .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); -- 1.9.1

3 2

[alsa-devel] [PATCH V6] ASoC: AMD: add AMD ASoC ACP-I2S driver
by Alex Deucher 21 Aug '15

21 Aug '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP IP block consists of dedicated DMA and I2S blocks. The PCM driver provides the DMA and CPU DAI components to ALSA core. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Reviewed-by: Murali Krishna Vemuri <murali-krishna.vemuri(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: - squash in Kconfig fix v3: - squash in the following patches: ASoC: AMD: add suspend/resume for ACP PCM driver ASoC: AMD: add ACP PCM driver runtime PM - Keep the Makefile sorted - Fix the Kconfig whitespace - drop RT286 changes to rework it using ACPI as a separate patch. The gpio stuff we need to handle this with ACPI was not added to the kernel until 4.2. This patch was originally developed much earlier. - put headers in include - remove static structure initialization - fix component name - remove casts from void - fix handling of combinations of play_intr and capture_intr - fix coding style with {} - switch to dev_ prints - remove substream checks handled by subsystem - fix whitespace in comments - pass return code directly on error from acp_dev->dma_start - drop references to designware - fix codeing style (STATUS_SUCCESS) - drop left over debugging messages - drop stuff that is already set in the dai_link - convert from using the amd_gnb_bus to an mfd cell v4: - change MODULE_ALIAS to platform:acp-i2s-audio v5: - failed attempt at a patch revision changelog v6: - respin the patch with a proper patch change log The module licensing follows the model used in the drm drivers. AMD prefers to contribute code under an MIT/X11 license. IANAL, but "GPL and additional rights" as the MODULE_LICENSE seems to be compatible with that and is exactly what most of the drm drivers do. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 5 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 676 ++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 686 insertions(+) create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 2ae9619..1b9ce31 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/adi/Kconfig" +source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e189903..a6cbb99 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += adi/ +obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig new file mode 100644 index 0000000..ce39979 --- /dev/null +++ b/sound/soc/amd/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_AMD_ACP + tristate "AMD Audio Coprocessor support" + depends on MFD_CORE + help + This option enables ACP support (DMA,I2S) on AMD platforms. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile new file mode 100644 index 0000000..1a66ec0 --- /dev/null +++ b/sound/soc/amd/Makefile @@ -0,0 +1,3 @@ +snd-soc-acp-pcm-objs := acp-pcm-dma.o + +obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c new file mode 100644 index 0000000..18cc60c --- /dev/null +++ b/sound/soc/amd/acp-pcm-dma.c @@ -0,0 +1,676 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2014-2015 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR + * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, + * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + * + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/amd_acp.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 2 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 2 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 1024 + +#define NUM_DSCRS_PER_CHANNEL 2 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + + +static const struct snd_pcm_hardware acp_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + /* formats,rates,channels based on i2s doc. */ + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + struct amd_acp_device *acp_dev; + struct acp_irq_prv *iprv; +}; + +static const struct snd_soc_component_driver dw_i2s_component = { + .name = "dw-i2s", +}; + +static void acp_pcm_period_elapsed(struct device *dev, u16 play_intr, + u16 capture_intr) +{ + struct snd_pcm_substream *substream; + struct audio_drv_data *irq_data = dev_get_drvdata(dev); + + /* Inform ALSA about the period elapsed (one out of two periods) */ + if (play_intr) + substream = irq_data->play_stream; + else + substream = irq_data->capture_stream; + + if (substream->runtime && snd_pcm_running(substream)) + snd_pcm_period_elapsed(substream); +} + +static int acp_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_drv_data *intr_data = dev_get_drvdata(prtd->platform->dev); + + struct audio_substream_data *adata = + kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); + if (adata == NULL) + return -ENOMEM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp_pcm_hardware_playback; + else + runtime->hw = acp_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(prtd->platform->dev, "set integer constraint failed\n"); + return ret; + } + + adata->acp_dev = intr_data->acp_dev; + runtime->private_data = adata; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + intr_data->play_stream = substream; + else + intr_data->capture_stream = substream; + + return 0; +} + +static int acp_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + uint64_t size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + u16 num_of_pages; + struct snd_pcm_runtime *runtime; + struct audio_substream_data *rtd; + struct amd_acp_device *acp_dev; + + dma_buffer = &substream->dma_buffer; + + runtime = substream->runtime; + rtd = runtime->private_data; + + if (WARN_ON(!rtd)) + return -EINVAL; + acp_dev = rtd->acp_dev; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + + if (NULL != pg) { + /* Save for runtime private data */ + rtd->pg = pg; + rtd->order = get_order(size); + + /* Let ACP know the Allocated memory */ + num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; + + /* Fill the page table entries in ACP SRAM */ + rtd->pg = pg; + rtd->size = size; + rtd->num_of_pages = num_of_pages; + rtd->direction = substream->stream; + + acp_dev->config_dma(acp_dev, rtd); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static int acp_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + pos = acp_dev->update_dma_pointer(acp_dev, substream->stream, + frames_to_bytes(runtime, runtime->period_size)); + return bytes_to_frames(runtime, pos); + +} + +static int acp_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp_dma_prepare(struct snd_pcm_substream *substream) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, ACP_TO_I2S_DMA_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH13, + NUM_DSCRS_PER_CHANNEL, 0); + /* Fill ACP SRAM (2 periods) with zeros from System RAM + * which is zero-ed in hw_params */ + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + ret = -EFAULT; + + /* Now configure DMA to transfer only first half of System RAM + * buffer before playback is triggered. This will overwrite + * zero-ed second half of SRAM buffer */ + acp_dev->config_dma_channel(acp_dev, SYSRAM_TO_ACP_CH_NUM, + PLAYBACK_START_DMA_DESCR_CH12, + 1, 0); + } else { + acp_dev->config_dma_channel(acp_dev, ACP_TO_SYSRAM_CH_NUM, + CAPTURE_START_DMA_DESCR_CH14, + NUM_DSCRS_PER_CHANNEL, 0); + acp_dev->config_dma_channel(acp_dev, I2S_TO_ACP_DMA_CH_NUM, + CAPTURE_START_DMA_DESCR_CH15, + NUM_DSCRS_PER_CHANNEL, 0); + } + return 0; +} + +static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + if (!rtd) + return -EINVAL; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_start(rtd->acp_dev, + SYSRAM_TO_ACP_CH_NUM, false); + if (ret < 0) + return -EFAULT; + acp_dev->prebuffer_audio(rtd->acp_dev); + + ret = acp_dev->dma_start(acp_dev, + ACP_TO_I2S_DMA_CH_NUM, true); + } else { + ret = acp_dev->dma_start(acp_dev, + I2S_TO_ACP_DMA_CH_NUM, true); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dev->dma_stop(acp_dev, SYSRAM_TO_ACP_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_I2S_DMA_CH_NUM); + } else { + ret = acp_dev->dma_stop(acp_dev, I2S_TO_ACP_DMA_CH_NUM); + if (0 == ret) + ret = acp_dev->dma_stop(acp_dev, + ACP_TO_SYSRAM_CH_NUM); + } + break; + default: + ret = -EINVAL; + + } + return ret; +} + +static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_pcm *pcm; + + pcm = rtd->pcm; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, MAX_BUFFER); + return ret; +} + +static int acp_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + + kfree(rtd); + + pm_runtime_mark_last_busy(prtd->platform->dev); + return 0; +} + +static int acp_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + struct device *dev = prtd->platform->dev; + u32 chan_nr; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + + default: + dev_err(dev, "unsuppted PCM fmt : %d\n", params_format(params)); + return -EINVAL; + } + + chan_nr = params_channels(params); + + switch (chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + rtd->ch_reg = 3; + break; + case SIX_CHANNEL_SUPPORT: + rtd->ch_reg = 2; + break; + case FOUR_CHANNEL_SUPPORT: + rtd->ch_reg = 1; + break; + case TWO_CHANNEL_SUPPORT: + rtd->ch_reg = 0; + break; + default: + dev_err(dev, "channel not supported : %d\n", chan_nr); + return -EINVAL; + } + + rtd->direction = substream->stream; + + acp_dev->config_i2s(acp_dev, rtd); + + return 0; +} + +static int acp_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + acp_dev->i2s_start(acp_dev, substream->stream); + ret = 0; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + acp_dev->i2s_stop(acp_dev, substream->stream); + ret = 0; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int acp_dai_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct audio_substream_data *rtd = runtime->private_data; + struct amd_acp_device *acp_dev = rtd->acp_dev; + + acp_dev->i2s_reset(acp_dev, substream->stream); + return 0; +} + +static struct snd_pcm_ops acp_dma_ops = { + .open = acp_dma_open, + .close = acp_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp_dma_hw_params, + .hw_free = acp_dma_hw_free, + .trigger = acp_dma_trigger, + .pointer = acp_dma_pointer, + .mmap = acp_dma_mmap, + .prepare = acp_dma_prepare, +}; + +static struct snd_soc_dai_ops acp_dai_i2s_ops = { + .prepare = acp_dai_i2s_prepare, + .hw_params = acp_dai_i2s_hwparams, + .trigger = acp_dai_i2s_trigger, +}; + +static struct snd_soc_dai_driver i2s_dai_driver = { + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp_dai_i2s_ops, +}; + +static struct snd_soc_platform_driver acp_asoc_platform = { + .ops = &acp_dma_ops, + .pcm_new = acp_dma_new, +}; + +static int acp_alsa_register(struct device *dev, struct amd_acp_device *acp_dev, + struct platform_device *pdev) +{ + int status; + + status = snd_soc_register_platform(dev, &acp_asoc_platform); + if (0 != status) { + dev_err(dev, "Unable to register ALSA platform device\n"); + goto exit_platform; + } else { + status = snd_soc_register_component(dev, + &dw_i2s_component, + &i2s_dai_driver, 1); + + if (0 != status) { + dev_err(dev, "Unable to register i2s dai\n"); + goto exit_dai; + } else { + dev_info(dev, "ACP device registered with ALSA\n"); + return status; + } + } + +exit_dai: + snd_soc_unregister_platform(dev); +exit_platform: + acp_dev->fini(acp_dev); + return status; +} + +static int acp_audio_probe(struct platform_device *pdev) +{ + int status; + struct audio_drv_data *audio_drv_data; + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), + GFP_KERNEL); + if (audio_drv_data == NULL) + return -ENOMEM; + + audio_drv_data->iprv = devm_kzalloc(&pdev->dev, + sizeof(struct acp_irq_prv), + GFP_KERNEL); + if (audio_drv_data->iprv == NULL) + return -ENOMEM; + + /* The following members gets populated in device 'open' + * function. Till then interrupts are disabled in 'acp_hw_init' + * and device doesn't generate any interrupts. + */ + + audio_drv_data->play_stream = NULL; + audio_drv_data->capture_stream = NULL; + audio_drv_data->acp_dev = acp_dev; + + audio_drv_data->iprv->dev = &pdev->dev; + audio_drv_data->iprv->acp_dev = acp_dev; + audio_drv_data->iprv->set_elapsed = acp_pcm_period_elapsed; + + dev_set_drvdata(&pdev->dev, audio_drv_data); + + /* Initialize the ACP */ + status = acp_dev->init(acp_dev, audio_drv_data->iprv); + + if (0 == status) + status = acp_alsa_register(&pdev->dev, acp_dev, pdev); + else + dev_err(&pdev->dev, "ACP initialization Failed\n"); + + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); + + return status; +} + +static int acp_audio_remove(struct platform_device *pdev) +{ + struct amd_acp_device *acp_dev = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + + acp_dev->fini(acp_dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static int acp_pcm_suspend(struct device *dev) +{ + bool pm_rts; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == false) + adata->acp_dev->fini(adata->acp_dev); + + return 0; +} + +static int acp_pcm_resume(struct device *dev) +{ + bool pm_rts; + struct snd_pcm_substream *pstream, *cstream; + struct snd_pcm_runtime *prtd, *crtd; + struct audio_substream_data *rtd; + struct audio_drv_data *adata = dev_get_drvdata(dev); + + pm_rts = pm_runtime_status_suspended(dev); + if (pm_rts == true) { + /* Resumed from system wide suspend and there is + * no pending audio activity to resume. */ + pm_runtime_disable(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + goto out; + } + + pstream = adata->play_stream; + prtd = pstream ? pstream->runtime : NULL; + if (prtd != NULL) { + /* Resume playback stream from a suspended state */ + rtd = prtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } + + cstream = adata->capture_stream; + crtd = cstream ? cstream->runtime : NULL; + if (crtd != NULL) { + /* Resume capture stream from a suspended state */ + rtd = crtd->private_data; + + adata->acp_dev->config_dma(adata->acp_dev, rtd); + adata->acp_dev->config_i2s(adata->acp_dev, rtd); + } +out: + return 0; +} + +static int acp_pcm_runtime_suspend(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_suspend(adata->acp_dev); + return 0; +} + +static int acp_pcm_runtime_resume(struct device *dev) +{ + struct audio_drv_data *adata = dev_get_drvdata(dev); + + adata->acp_dev->acp_resume(adata->acp_dev); + return 0; +} + +static const struct dev_pm_ops acp_pm_ops = { + .suspend = acp_pcm_suspend, + .resume = acp_pcm_resume, + .runtime_suspend = acp_pcm_runtime_suspend, + .runtime_resume = acp_pcm_runtime_resume, +}; + +static struct platform_driver acp_dma_driver = { + .probe = acp_audio_probe, + .remove = acp_audio_remove, + .driver = { + .name = "acp-i2s-audio", + .pm = &acp_pm_ops, + }, +}; + +module_platform_driver(acp_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP PCM Driver"); +MODULE_LICENSE("GPL and additional rights"); +MODULE_ALIAS("platform:acp-i2s-audio"); -- 1.8.3.1

1 0

[alsa-devel] Boss Micro BR-80 not workin
by Karsten Heymann 20 Aug '15

20 Aug '15

Hi, I'm trying to get my Boss Micro BR-80 (id 0582:0130) usb device to work. The device is detected, but with my limited knowledge of the alsa basics i could not get any audio in or out of it, pulse audio and jack don't seem to work too. I'm not a kernel hacker, but hope to be helpful with providing output, testing patches and if necessary providing ssh access to a system with this device attached to a more knowledgeable developer. There used to be dedicated quirk code in sound/usb/quirk-table.h which was removed in http://thread.gmane.org/gmane.linux.alsa.devel/109660/focus=111768 during a roland module update. Older postings on the net indicate that the br-80 seem to have worked in the past. My questions: * which command outputs do you need to debug this? * should compiling an older version of the snd-usb module for a current kernel work? Regards, Karsten

1 0

[alsa-devel] [PATCH] ASoC: wm8994: fix add dynamic path error
by Zidan Wang 20 Aug '15

20 Aug '15

If there don't exist dynamic sink or source widget, it will failed to add dynamic path. "AIF3ADCDAT" is snd_soc_dapm_aif_out, can't be dynamic sink widget. So change the audio route to fix this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8994.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 962e1d3..2ccbb32 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1942,14 +1942,16 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACR" }, + + { "AIF3ADCDAT", NULL, "AIF3ADC Mux" }, /* Loopback */ { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: ac97: Switch to dev_pm_ops
by Lars-Peter Clausen 20 Aug '15

20 Aug '15

Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single AC'97 driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch touches code in both ALSA and the input subsystem. Ideally this patch will be merged via the ALSA tree. Given that the wm97xx touchscreen driver is not seeing too many changes these days the risk of conflicts should be low. --- drivers/input/touchscreen/wm97xx-core.c | 13 +++++++------ sound/ac97_bus.c | 26 -------------------------- 2 files changed, 7 insertions(+), 32 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index b1ae779..1d11fd8 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -732,8 +732,8 @@ static int wm97xx_remove(struct device *dev) return 0; } -#ifdef CONFIG_PM -static int wm97xx_suspend(struct device *dev, pm_message_t state) +#ifdef CONFIG_PM_SLEEP +static int wm97xx_suspend(struct device *dev) { struct wm97xx *wm = dev_get_drvdata(dev); u16 reg; @@ -799,9 +799,11 @@ static int wm97xx_resume(struct device *dev) return 0; } +static SIMPLE_DEV_PM_OPS(wm97xx_pm_ops, wm97xx_suspend, wm97xx_resume); +#define WM97XX_PM_OPS (&wm97xx_pm_ops) + #else -#define wm97xx_suspend NULL -#define wm97xx_resume NULL +#define WM97XX_PM_OPS NULL #endif /* @@ -836,8 +838,7 @@ static struct device_driver wm97xx_driver = { .owner = THIS_MODULE, .probe = wm97xx_probe, .remove = wm97xx_remove, - .suspend = wm97xx_suspend, - .resume = wm97xx_resume, + .pm = WM97XX_PM_OPS, }; static int __init wm97xx_init(void) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 2b50cbe..57a6dfc 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -27,35 +27,9 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return 1; } -#ifdef CONFIG_PM -static int ac97_bus_suspend(struct device *dev, pm_message_t state) -{ - int ret = 0; - - if (dev->driver && dev->driver->suspend) - ret = dev->driver->suspend(dev, state); - - return ret; -} - -static int ac97_bus_resume(struct device *dev) -{ - int ret = 0; - - if (dev->driver && dev->driver->resume) - ret = dev->driver->resume(dev); - - return ret; -} -#endif /* CONFIG_PM */ - struct bus_type ac97_bus_type = { .name = "ac97", .match = ac97_bus_match, -#ifdef CONFIG_PM - .suspend = ac97_bus_suspend, - .resume = ac97_bus_resume, -#endif /* CONFIG_PM */ }; static int __init ac97_bus_init(void) -- 2.1.4

3 10

[alsa-devel] [GIT PULL] ASoC update for v4.2
by Mark Brown 19 Aug '15

19 Aug '15

The following changes since commit bc0195aad0daa2ad5b0d76cce22b167bc3435590: Linux 4.2-rc2 (2015-07-12 15:10:30 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2-disable-topology for you to fetch changes up to 8c90503bf246bebb48caa5590d41df755ba08550: ASoC: topology: Disable use from userspace (2015-08-18 09:33:21 -0700) ---------------------------------------------------------------- ASoC: Disable topology support for v4.2 The topology code merged in the v4.2 merge window introduced a new ABI which was believed to be suitable for use but subsequently additional work by the developers of this feature have revealed some problems that need to be addressed. In order to allow this to be done without having to support the initial ABI add Kconfig to disable the build and also add some #error statements to the UAPI header so users can't use them. ---------------------------------------------------------------- Mark Brown (2): ASoC: topology: Add Kconfig option for topology ASoC: topology: Disable use from userspace include/sound/soc-topology.h | 12 ++++++++++++ include/uapi/sound/asoc.h | 6 ++++++ sound/soc/Kconfig | 3 +++ sound/soc/Makefile | 3 +++ 4 files changed, 24 insertions(+)

2 1

[alsa-devel] [PATCH v3] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 19 Aug '15

19 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V3: Fixed MODULE_LICENSE string. V2: Update after comments from Lars-Peter: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. I've left the checks in dai_set_fmt, as (IIUC) there currently doesn't seem to be a way for the ALSA framework to handle format constraints using the codec configuration struct. .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPLv2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Adjust devm usage
by Vaishali Thakkar 19 Aug '15

19 Aug '15

Remove use of snd_soc_unregister_component in remove function as devm_snd_soc_register_component in probe function automatically handles it. Also, convert call of snd_dmaengine_pcm_register to managed resource function devm_snd_dmaengine_pcm_register and remove usage of snd_dmaengine_pcm_unregister in probe and remove functions. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index acb5be5..b936102 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -483,16 +483,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + return ret; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) i2s_runtime_suspend(&pdev->dev); @@ -512,8 +510,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) clk_disable_unprepare(i2s->mclk); clk_disable_unprepare(i2s->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix gain settings of FLL in free-run mode
by Charles Keepax 19 Aug '15

19 Aug '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> The wrong register was used to set the gain of ref loop, when changing the FLL output on an active FLL. This patch corrects the offset of the gain register. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e759d39..42b2021 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2058,7 +2058,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Recurse before saving terminal properties
by Julian Scheel 19 Aug '15

19 Aug '15

The input terminal parser recurses into the referenced clock entity to verify it is existant and thus the terminal descriptor is valid. The actual property values of the term instance which is initially parsed must not be overriden by the recursion. For this to work the term properties have to be assigned after recursing into the referenced clock entity descriptors. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Store term->id after recursion as well, as it gets overriden in the recursion - Add comments explaining that the recursion is only for validation of the descriptor validity --- sound/usb/mixer.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81055d3..c50790c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -731,15 +731,21 @@ static int check_input_term(struct mixer_build *state, int id, term->name = d->iTerminal; } else { /* UAC_VERSION_2 */ struct uac2_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - /* call recursively to get the clock selectors */ + /* call recursively to verify that the + * referenced clock entity is valid */ err = check_input_term(state, d->bCSourceID, term); if (err < 0) return err; + + /* save input term properties after recursion, + * to ensure they are not overriden by the + * recursion calls */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; } return 0; case UAC_FEATURE_UNIT: { -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoc: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 19 Aug '15

19 Aug '15

(Referring to this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html) This patch adds the capability to use multiple codecs on the same DAI link where one codec is used for playback and another one is used for capture. As noted in the thread, there might be more places where the new snd_soc_dai_stream_valid() function is needed, however, these two were sufficient for our use case (SSM2518 playback and ICS-43432 capture). Signed-off-by: Ricard Wanderlof <ricardw(a)xis.com> --- sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e..df1f261 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,32 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * @dai: The DAI to check for stream support + * @stream: The stream type (playback / capture) to check for + * + * Checks if the DAI in question supports the indicated stream type. + * Needed to skip over codecs when iterating over + * snd_soc_pcm_runtime->codec_dais in order to skip codecs that are playback- + * or capture-only, which would otherwise inhibit capture or playback, + * respectively, to the other codec in a multiple codec / common I2S bus + * environment. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return !!codec_stream->rates; +} + +/** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated * @stream: Direction of the PCM stream @@ -371,6 +397,12 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 4

[alsa-devel] [PATCH v3 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 19 Aug '15

19 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Changes in v3: - Fix typos in commit message - Resent patches to a more complete list of maintainers. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 381 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 536 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.5.0

3 9

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad W541 (17aa:2211)
by Rick Sherman 19 Aug '15

19 Aug '15

Tested with W541 and Ultra Dock 170w Signed-off-by: Rick Sherman <rick(a)shermdog.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b9847a..39a0342 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5291,6 +5291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.4.3

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix runtime PM unbalance
by Takashi Iwai 19 Aug '15

19 Aug '15

The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio: Fix deadlocks at resuming] introduced a new check of in_pm flag. However, the brainless patch author evaluated it in a wrong way (logical AND instead of logical OR), thus usb_autopm_get_interface() is never called, leading to unbalance of runtime PM refcount. This patch fixes it by correcting the logic. Reported-by: Hans Yang <hansy(a)nvidia.com> Fixes: 1ee23fe07ee8 ('ALSA: usb-audio: Fix deadlocks at resuming') Cc: <stable(a)vger.kernel.org> [v3.15+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1fab9778807a..0450593980fd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -638,7 +638,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) int err = -ENODEV; down_read(&chip->shutdown_rwsem); - if (chip->probing && chip->in_pm) + if (chip->probing || chip->in_pm) err = 0; else if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: wm8741: Drop misleading readable_reg callback implementation
by Axel Lin 18 Aug '15

18 Aug '15

regmap_readable() returns false if map->format.format_write is set. For .reg_bits = 7, .val_bits = 9, setting, map->format.format_write = regmap_format_7_9_write; Even current code has implemented map->readable_reg, regmap_readable() still returns false anyway. Thus drop the misleading readable_reg callback implementation. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9723e44..de42c03 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -61,25 +61,6 @@ static const struct reg_default wm8741_reg_defaults[] = { { 32, 0x0002 }, /* R32 - ADDITONAL_CONTROL_1 */ }; -static bool wm8741_readable(struct device *dev, unsigned int reg) -{ - switch (reg) { - case WM8741_DACLLSB_ATTENUATION: - case WM8741_DACLMSB_ATTENUATION: - case WM8741_DACRLSB_ATTENUATION: - case WM8741_DACRMSB_ATTENUATION: - case WM8741_VOLUME_CONTROL: - case WM8741_FORMAT_CONTROL: - case WM8741_FILTER_CONTROL: - case WM8741_MODE_CONTROL_1: - case WM8741_MODE_CONTROL_2: - case WM8741_ADDITIONAL_CONTROL_1: - return true; - default: - return false; - } -} - static int wm8741_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8741_RESET, 0); @@ -541,8 +522,6 @@ static const struct regmap_config wm8741_regmap = { .reg_defaults = wm8741_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8741_reg_defaults), .cache_type = REGCACHE_RBTREE, - - .readable_reg = wm8741_readable, }; static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: topology: Disable use from userspace
by Mark Brown 18 Aug '15

18 Aug '15

Since the topology API is still in sufficient flux for changes to be identified disable the use of the userspace ABI by adding #error statements to the code, ensuring that nobody relies on the headers as currently defined. It is expected that this change will be reverted for v4.3. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 51b8066..247c50b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,6 +18,12 @@ #include <linux/types.h> #include <sound/asound.h> +#ifndef __KERNEL__ +#error This API is an early revision and not enabled in the current +#error kernel release, it will be enabled in a future kernel version +#error with incompatible changes to what is here. +#endif + /* * Maximum number of channels topology kcontrol can represent. */ -- 2.5.0

1 1

[alsa-devel] [PATCH] ASoC: tegra: Convert to managed resources
by Vaishali Thakkar 18 Aug '15

18 Aug '15

Use managed resource functions devm_clk_put and devm_snd_soc_register_component to simplify error handling. To be compatible with the change various gotos are replaced with direct returns, and unneeded labels are dropped. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 37 +++++++++++++------------------------ 1 file changed, 13 insertions(+), 24 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 9141477..f69b2e4 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -273,45 +273,40 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) GFP_KERNEL); if (!spdif) { dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } dev_set_drvdata(&pdev->dev, spdif); - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + spdif->clk_spdif_out = devm_clk_get(&pdev->dev, "spdif_out"); if (IS_ERR(spdif->clk_spdif_out)) { pr_err("Can't retrieve spdif clock\n"); ret = PTR_ERR(spdif->clk_spdif_out); - goto err; + return ret; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } memregion = devm_request_mem_region(&pdev->dev, mem->start, resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; + return -EBUSY; } regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; + return -ENOMEM; } spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, @@ -319,7 +314,7 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); ret = PTR_ERR(spdif->regmap); - goto err_clk_put; + return ret; } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; @@ -334,8 +329,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, - &tegra20_spdif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -345,21 +341,17 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; + return ret; } return 0; -err_unregister_component: - snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err: + return ret; } @@ -372,9 +364,6 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - clk_put(spdif->clk_spdif_out); return 0; } -- 1.9.1

3 8

[alsa-devel] Segmentation Fault in snd_pcm_rate_hw_free()
by Alexandru Costache 18 Aug '15

18 Aug '15

Hi, I was reproducing the crash that Valentin Corfu reported earlier with his test. Valgrind and gdb show that expand_linear_s16() is somehow corrupting the area used for memory management by libc for rate->pareas[0].addr More exactly, the line that seems to corrupt the heap in static void expand_linear_s16() is: *dst = (old_sample * old_weight + new_sample * new_weight) >> 16; As Valgrind says: ==11161== Invalid write of size 2 ==11161== at 0x40A8A82: linear_expand_s16 (pcm_rate_linear.c:179) ==11161== by 0x40A85A9: linear_convert (pcm_rate_linear.c:320) ==11161== by 0x40A6915: do_convert (pcm_rate.c:537) ==11161== by 0x40A6C7D: snd_pcm_rate_write_areas1 (pcm_rate.c:550) ==11161== by 0x40A6C7D: snd_pcm_rate_commit_area (pcm_rate.c:749) ==11161== by 0x40A7014: snd_pcm_rate_drain (pcm_rate.c:1105) ==11161== by 0x4084511: snd_pcm_drain (pcm.c:1122) ==11161== by 0x8049288: main (in /root/pcm_min_daisuke) ==11161== Address 0x43d6258 is 0 bytes after a block of size 86,016 alloc'd ==11161== at 0x40261B1: malloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x4099801: snd_pcm_mmap (pcm_mmap.c:425) ==11161== by 0x4093CA7: sndrv_pcm_hw_params (pcm_params.c:2366) ==11161== by 0x409A59C: snd1_pcm_generic_hw_params (pcm_generic.c:104) ==11161== by 0x4091E7F: snd1_pcm_hw_params_slave (pcm_params.c:2250) ==11161== by 0x40A54AC: snd_pcm_rate_hw_params (pcm_rate.c:246) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x40ABC37: snd_pcm_plug_hw_params (pcm_plug.c:1045) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x4084281: snd_pcm_hw_params (pcm.c:830) ==11161== by 0x8048F84: main (in /root/pcm_min_daisuke) ==11161== ==11161== Invalid free() / delete / delete[] / realloc() ==11161== at 0x402726D: free (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x40A53C0: snd_pcm_rate_hw_free (pcm_rate.c:344) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x40A9E8D: snd_pcm_plug_hw_free (pcm_plug.c:1061) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x80492AC: main (in /root/pcm_min_daisuke) ==11161== Address 0xe705004b is not stack'd, malloc'd or (recently) free'd I'm not very familiar with alsa in general and was wondering if someone who knows more on it can drop a look on this one? Thank you, Alexandru

2 1

[alsa-devel] [PATCH v4 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 18 Aug '15

18 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

1 3

[alsa-devel] [PATCH] ASoC: topology: Add Kconfig option for topology
by Mark Brown 18 Aug '15

18 Aug '15

Allow the topology code to be compiled out so that users who don't need topology don't need to havve the code compiled in, saving them some memory. Some more configuration could be added to remove some of the hooks into the core data structures but that is probably best done with some refactoring to use functions to do the updates of the data structures rather than ifdefing in the code as we'd need to do at the minute. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 12 ++++++++++++ sound/soc/Kconfig | 3 +++ sound/soc/Makefile | 3 +++ 3 files changed, 18 insertions(+) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..427bc41 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -141,6 +141,8 @@ struct snd_soc_tplg_ops { int io_ops_count; }; +#ifdef CONFIG_SND_SOC_TOPOLOGY + /* gets a pointer to data from the firmware block header */ static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) { @@ -165,4 +167,14 @@ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, int num_events, u16 event_type); +#else + +static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp, + u32 index) +{ + return 0; +} + +#endif + #endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 467ee1a..225bfda 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -30,6 +30,9 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_TOPOLOGY + bool + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 19fd711..134aca1 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o + +ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o +endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 2.5.0

2 4

[alsa-devel] [PATCH v2 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 18 Aug '15

18 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list ASoC: Add support for dummy DAI links and PCM runtimes ASoC: Bind new DAI links after probing components ASoC: Support adding a DAI dynamically ASoC: topology: Change pass number of DAI smaller than graph ASoC: topology: Change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

6 33

Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Takashi Iwai 18 Aug '15

18 Aug '15

On Tue, 18 Aug 2015 01:03:31 +0200, Paulo Roberto de Oliveira Castro wrote: > > Hello again, > > I am using Arch now, and today kernel 4.1.5 was released on Arch's repo. > However, the sound isn't working yet. Am I missing something? The patch is included in 4.1.5. Make sure that you tested really properly. If not, check again whether 4.2 works in the same condition. Takashi > > 2015-07-29 14:40 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > On Wed, 29 Jul 2015 18:48:29 +0200, > > Paulo Roberto de Oliveira Castro wrote: > > > > > > Thanks! Is there a way to track these changes? > > > > Track git tree :) > > > > At first, it's found in my sound git tree, then it'll be merged to > > Linus git tree, and then to each stable tree. > > > > > > Takashi > > > > > > > > 2015-07-29 4:08 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > > > > > On Wed, 29 Jul 2015 00:32:08 +0200, > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > I'm not an expert on applying patches to the kernel (never did that > > > > > before), therefore, I don't know if I did everything right. > > > > > > > > > > What I did was: downloaded 4.2-rc4 from kernel.org[1], added the > > line by > > > > > hand (couldn't apply the patch using patch), copied the config file > > from > > > > > /boot/ (the config fedora 22 uses for its kernel) to .config on the > > root > > > > of > > > > > the kernel directory, runned "make oldconfig" (typing ENTER for every > > > > > question, is that right?), "make", "make modules_install", "make > > > > install". > > > > > Rebooted, and chose the new kernel I just installed. > > > > > > > > > > The sound worked like a charm, the speakers worked as expected, and > > the > > > > > volume was as loud as on Windows. > > > > > > > > Good to hear, I queued the fix now. > > > > > > > > > Is the patch going to be applied on master? Any chance that fix > > reaches > > > > my > > > > > Fedora 22 (with backport to older kernel versions)? > > > > > > > > The commit is with Cc to stable, so it'll be backported to stable > > > > kernels once when landed to upstream. Then Fedora will pick it up > > > > eventually later. > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > > > > > > > Thanks for the help, > > > > > > > > > > Paulo > > > > > > > > > > [1] > > > > > > > > > > > https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz > > > > > > > > > > 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < > > > > > p.oliveira.castro(a)gmail.com>: > > > > > > > > > > > > > > > > > > > > > > > ---------- Forwarded message --------- > > > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > > > Date: Tue, Jul 28, 2015, 03:35 > > > > > > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) > > > > kernel > > > > > > driver > > > > > > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > > > > > > Cc: <alsa-devel(a)alsa-project.org> > > > > > > > > > > > > > > > > > > On Tue, 28 Jul 2015 05:35:34 +0200, > > > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > > > > > Hi everyone, > > > > > > > > > > > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers > > don't > > > > > > work, > > > > > > > and the sound only comes out of the subwoofer. Here is the > > output of > > > > > > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > > > > > > workaround: > > > > > > > > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # > > > > enable, > > > > > > > set it to OUT > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to > > 0x0c > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # > > unmute > > > > left > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # > > unmute > > > > > > right > > > > > > > > > > > > > > We suspect that it's a kernel driver bug. This makes the speakers > > > > work > > > > > > > alongside the headphones (both at the same time) but I need help > > > > making > > > > > > it > > > > > > > working when headphones are not plugged in. > > > > > > > > > > > > Does the patch below work? Try with 4.2-rc kernel at best. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > --- > > > > > > diff --git a/sound/pci/hda/patch_realtek.c > > > > b/sound/pci/hda/patch_realtek.c > > > > > > index 4ae877c3b6a1..18ae17ebb356 100644 > > > > > > --- a/sound/pci/hda/patch_realtek.c > > > > > > +++ b/sound/pci/hda/patch_realtek.c > > > > > > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk > > > > alc269_fixup_tbl[] > > > > > > = { > > > > > > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > > > > > > ALC288_FIXUP_DELL_XPS_13), > > > > > > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > > > > > > ALC290_FIXUP_SUBWOOFER_HSJACK), > > > > > > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > > > > > > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > > > > > > _______________________________________________ > > > > > Alsa-devel mailing list > > > > > Alsa-devel(a)alsa-project.org > > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > >

1 0

[alsa-devel] [PATCH] ASoC: davinci-vcif: Use devm_snd_soc_register_component
by Vaishali Thakkar 17 Aug '15

17 Aug '15

Use resource managed function devm_snd_soc_register_component for component registration instead of snd_soc_register_component. Also, remove davinci_vcif_remove as it is now redundant. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fabd05f..c77d921 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -231,8 +231,9 @@ static int davinci_vcif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, - &davinci_vcif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &davinci_vcif_component, + &davinci_vcif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); return ret; @@ -241,23 +242,14 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - snd_soc_unregister_component(&pdev->dev); return ret; } return 0; } -static int davinci_vcif_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; -} - static struct platform_driver davinci_vcif_driver = { .probe = davinci_vcif_probe, - .remove = davinci_vcif_remove, .driver = { .name = "davinci-vcif", }, -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Remove AIF widgets for backend DAIs
by Koro Chen 17 Aug '15

17 Aug '15

DAPM core already creates widgets for DAIs. It is not necessary to declare them by SND_SOC_DAPM_AIF_IN/SND_SOC_DAPM_AIF_OUT. Furthermore, original codes use backend DAI's stream name to be the AIF widget name. It causes the same widget to be created twice, and after commit 92fa12426741 ("ASoC: dapm: Add new widgets to the end of the widget list") the first created widget (by snd_soc_dapm_new_controls) is used, not the 2nd created one (by snd_soc_dapm_new_dai_widgets), so audio path is broken. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 55471ee..d190fe0 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -897,10 +897,6 @@ static const struct snd_kcontrol_new mtk_afe_o10_mix[] = { }; static const struct snd_soc_dapm_widget mtk_afe_pcm_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_IN("I2S Capture", NULL, 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT("I2S Playback", NULL, 0, SND_SOC_NOPM, 0, 0), - /* inter-connections */ SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -932,11 +928,6 @@ static const struct snd_soc_dapm_route mtk_afe_pcm_routes[] = { { "O10", "I18 Switch", "I18" }, }; -static const struct snd_soc_dapm_widget mtk_afe_hdmi_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_OUT("HDMIO Playback", NULL, 0, SND_SOC_NOPM, 0, 0), -}; - static const struct snd_soc_dapm_route mtk_afe_hdmi_routes[] = { {"HDMIO Playback", NULL, "HDMI"}, }; @@ -951,8 +942,6 @@ static const struct snd_soc_component_driver mtk_afe_pcm_dai_component = { static const struct snd_soc_component_driver mtk_afe_hdmi_dai_component = { .name = "mtk-afe-hdmi-dai", - .dapm_widgets = mtk_afe_hdmi_widgets, - .num_dapm_widgets = ARRAY_SIZE(mtk_afe_hdmi_widgets), .dapm_routes = mtk_afe_hdmi_routes, .num_dapm_routes = ARRAY_SIZE(mtk_afe_hdmi_routes), }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH RFC v3 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 17 Aug '15

17 Aug '15

This is my third RFC patch series on the subject. This time also the tda998x driver patches are a serious attempt to come up with an initial implementation for HDMI audio ASoC support. Russell's "drm/edid: add function to help find SADs" is here just to produce a working patch-set on top of mainline. It should be getting in trough drm-next at some point. The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (5): ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support Russell King - ARM Linux (1): drm/edid: add function to help find SADs, DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 336 +++++++++++++-- include/drm/drm_edid.h | 19 + include/sound/hdmi-codec.h | 99 +++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 467 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ----- 10 files changed, 1026 insertions(+), 153 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 17 Aug '15

17 Aug '15

V2: Update after comments from Lars-Peter yesterday: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v2] ASoC: ssm2518: Add explicit device tree support
by Ricard Wanderlof 17 Aug '15

17 Aug '15

Add OF match table to SSM2518 to allow direct matching without going through I2C subsystem. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..2d199b1 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +811,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +829,7 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ssm2518_dt_ids), }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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Re: [alsa-devel] probe issue for Dice/OXFW drivers
by Clemens Ladisch 17 Aug '15

17 Aug '15

Takashi Sakamoto wrote: > Recently, I received an issue from a user of Mackie Onyx-i series. > According to the user, snd-dice is not applied to his Dice model. > [...] According to the log, I cannot find the first > transaction which snd-dice performs. Additionally, no error messages in > syslog. There are some checks performed before the first actual transaction. The kernel loads the driver if an entry in dice_id_table[] matches, i.e., if version==1. This values indeed is in the unit directory: > ROM header and bus information block > ----------------------------------------------------------------- > 400 0404f507 bus_info_length 4, crc_length 4, crc 62727 > 404 31333934 bus_name "1394" > 408 e0008102 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512) > 40c 000ff210 company_id 000ff2 | > 410 01800af7 device_id 1001800af7 | EUI-64 000ff21001800af7 > > root directory > ----------------------------------------------------------------- > 414 000639d3 directory_length 6, crc 14803 > 418 03000ff2 vendor > 41c 8100000a --> descriptor leaf at 444 > 420 17000006 model > 424 8100000d --> descriptor leaf at 458 > 428 0c0087c0 node capabilities per IEEE 1394 > 42c d1000001 --> unit directory at 430 > > unit directory at 430 > ----------------------------------------------------------------- > 430 00045246 directory_length 4, crc 21062 > 434 12000ff2 specifier id > 438 13000001 version > 43c 17000006 model > 440 8100000b --> descriptor leaf at 46c The dice_interface_check() function then makes other checks: /* * Check that GUID and unit directory are constructed according to DICE * rules, i.e., that the specifier ID is the GUID's OUI, and that the * GUID chip ID consists of the 8-bit category ID, the 10-bit product * ID, and a 22-bit serial number. */ These values are correct, except for the category ID. The driver already has an category ID exception for Weiss devices. It would be possible to add another exception, but given that firmware writers apparently like to change this value without good reason, it might be a better idea to just remove the category ID check. Regards, Clemens

2 1

[alsa-devel] [PATCH 00/25] ALSA: support AMDTP variants
by Takashi Sakamoto 17 Aug '15

17 Aug '15

This patchset updates my previous RFC, excluding control/status message for TASCAM FireWire series and a MOTU driver, [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094789.html Unfortunately, I've got no dedicated testers to this patchset. While, the supported models have little time to live. I think it's time to merge them. This patchset supports playbacking/capturing PCM samples/MIDI messages. Model specific functionalities such as MIDI mapping to physical controls should be done in userspace, with propper asynchronous transactions. I hope users to remind that my work is done by reasoning and assuming with a result of investigating packets on IEEE 1394 bus. There's no published specification, thus these drivers cannot fully satisfies users' expectation which comes from Windows or OS X. Changes: * Fix minor bugs, mainly for asynchronous transaction helper for MIDI port. * Regain PCM hardware constraint (I missed it). * Fix TASCAM quirk for MIDI transactions, as much as possible. Rest of work (but may not be done in this developing period): * Handling control/status messages for TASCAM driver. * Supporting MIDI exclusive message for outgoing packets. * Researching a reason to de-synchronize for Digi00x family. * Improving MIDI throttle for asynchronous transaction helper Takashi Sakamoto (25): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation for packet transmission layer and data processing layer ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 9 + sound/firewire/Kconfig | 27 ++ sound/firewire/Makefile | 4 +- sound/firewire/amdtp-am824.c | 451 ++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 47 +++ sound/firewire/{amdtp.c => amdtp-stream.c} | 375 ++++---------------- sound/firewire/{amdtp.h => amdtp-stream.h} | 113 ++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 41 ++- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 436 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 353 +++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 ++++++ sound/firewire/digi00x/digi00x-stream.c | 422 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++++++ sound/firewire/digi00x/digi00x.c | 174 ++++++++++ sound/firewire/digi00x/digi00x.h | 157 +++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 129 +++++++ sound/firewire/lib.h | 52 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 248 +++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 202 +++++++++++ sound/firewire/tascam/tascam-midi.c | 162 +++++++++ sound/firewire/tascam/tascam-pcm.c | 298 ++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 +++++ sound/firewire/tascam/tascam-stream.c | 497 +++++++++++++++++++++++++++ sound/firewire/tascam/tascam-transaction.c | 291 ++++++++++++++++ sound/firewire/tascam/tascam.c | 230 +++++++++++++ sound/firewire/tascam/tascam.h | 140 ++++++++ 48 files changed, 5277 insertions(+), 498 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (77%) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

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[alsa-devel] [PATCH 0/4] Minor cleanups to hda code
by Takashi Iwai 17 Aug '15

17 Aug '15

Hi, this is a series of trivial cleanup patches for HDA driver, mostly for procfs code. Takashi Iwai (4): ALSA: hda - Make some helper functions local ALSA: hda/proc - Add const to possible places ALSA: hda/proc - Fix racy string access for power states ALSA: hda/eld - Add const to possible places sound/pci/hda/hda_codec.c | 70 ------------------------------ sound/pci/hda/hda_codec.h | 7 --- sound/pci/hda/hda_eld.c | 10 ++--- sound/pci/hda/hda_proc.c | 107 +++++++++++++++++++++++++++++++++++----------- 4 files changed, 86 insertions(+), 108 deletions(-) -- 2.5.0

1 4

[alsa-devel] [PATCH] ASoC: topology: Add support for TLV bytes control
by mengdong.lin＠intel.com 17 Aug '15

17 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Allow vendor drivers to define bespoke bytes ext handlers and IDs for TLV bytes controls. And the topology core will bind these handlers by matching IDs defined by the vendor driver and user space topology data file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..3e412c6 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -89,6 +89,13 @@ struct snd_soc_tplg_kcontrol_ops { struct snd_ctl_elem_info *uinfo); }; +/* Bytes ext operations, for TLV byte controls */ +struct snd_soc_tplg_bytes_ext_ops { + u32 id; + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); +}; + /* * DAPM widget event handlers - used to map handlers onto widgets. */ @@ -139,6 +146,10 @@ struct snd_soc_tplg_ops { /* bespoke kcontrol handlers available for binding */ const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + + /* bespoke bytes ext handlers available for binding */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; }; /* gets a pointer to data from the firmware block header */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 607f98b..b5a2868 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -70,6 +70,10 @@ struct soc_tplg { const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + /* bespoke bytes ext handlers, for TLV bytes controls */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; + /* optional fw loading callbacks to component drivers */ struct snd_soc_tplg_ops *ops; }; @@ -513,6 +517,15 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, { int i; + /* TLV bytes controls don't need get/put ops */ + if (k->iface & SNDRV_CTL_ELEM_IFACE_MIXER + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK + && hdr->ops.info == SND_SOC_TPLG_CTL_BYTES) { + k->info = snd_soc_bytes_info_ext; + return 0; + } + /* try and map standard kcontrols handler first */ for (i = 0; i < num_ops; i++) { @@ -547,6 +560,32 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, return -EINVAL; } +/* bind bytes ext get/put handlers for TLV bytes controls */ +static int soc_tplg_bytes_ext_bind_io(struct soc_tplg *tplg, + struct snd_soc_tplg_io_ops *ops_id, + struct soc_bytes_ext *sbe) +{ + const struct snd_soc_tplg_bytes_ext_ops *ops = tplg->bytes_ext_ops; + int num_ops = tplg->bytes_ext_ops_count; + int i; + + /* try bespoke handlers */ + for (i = 0; i < num_ops; i++) { + + if (sbe->put == NULL && ops[i].id == ops_id->put) + sbe->put = ops[i].put; + if (sbe->get == NULL && ops[i].id == ops_id->get) + sbe->get = ops[i].get; + } + + /* bespoke handlers found ? */ + if (sbe->put && sbe->get) + return 0; + + /* nothing to bind */ + return -EINVAL; +} + /* bind a widgets to it's evnt handlers */ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, @@ -690,6 +729,14 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc.access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, (struct snd_soc_tplg_ctl_hdr *)be); @@ -1315,6 +1362,14 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc[i].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], (struct snd_soc_tplg_ctl_hdr *)be); @@ -1736,6 +1791,8 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.req_index = id; tplg.io_ops = ops->io_ops; tplg.io_ops_count = ops->io_ops_count; + tplg.bytes_ext_ops = ops->bytes_ext_ops; + tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; return soc_tplg_load(&tplg); } -- 1.9.1

3 5

[alsa-devel] [PATCH v2 0/2] USB audio fixes
by Pierre-Louis Bossart 16 Aug '15

16 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. v2: applied Takashi's comments unconditional clean-up of sync_endpoint pointers better logical tests and removal of goto Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 3

[alsa-devel] BugReport: Dice works partially [Broken]
by Stanislav Sergiienko 16 Aug '15

16 Aug '15

Hi, I am a user of TC Electronic DesktopKonnekt6 and latest kernel update 3.19.2 add partial support for my device and thus broke my usual sound workflow. So i want to make good bug report to make device usable on latest kernels. With 3.19.2 kernel device is successfully recognized by system and even sometimes can play sound(!). But always, after some time sound is breaks (usually after 30-90 seconds) and to restore it i must restart pulseaudio(v6.0) and application. So device becomes unusable on latests kernels. Earlier i use my sound card through ffado + jackdbus + pulseaudio. Now it's impossible, so i have to downgrade kernel. Please request any additional info that you need to fix problem and i will post it here soon. Also i can test any "bleeding edge" changes thru dkms as described here: https://github.com/takaswie/snd-firewire-improve Related bug(not mine): https://bugs.archlinux.org/task/44258 Some useful info: cat /proc/asound/cards 0 [DesktopKonnekt6]: DICE - DesktopKonnekt6 TC Electronic DesktopKonnekt6 (serial 12941) at fw1.0, S400 cat /proc/asound/card0/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc1:000100000000 notification: 00000040 nick name: DesktopKonnekt6 clock select: internal 44100 enable: 1 status: locked 44100 ext status: 00000000 sample rate: 44100 version: 1.0.4.0 clock caps: 44100 48000 88200 96000 176400 192000 arx1 arx2 internal clock source names: Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\INTERNAL\\ tx 0: iso channel: 0 audio channels: 6 midi ports: 0 speed: S400 names: ch 1\ch 2\ - \ - \ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 rx 0: iso channel: 1 sequence start: 0 audio channels: 6 midi ports: 0 names: main out 1\main out 2\phones out 1\phones out 2\ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 ext status: clock source: internal locked: 1 rate: 44100 adat user data: 0

3 4

[alsa-devel] [PATCH 0/9] Add DSP topology management for SKL
by Subhransu S. Prusty 15 Aug '15

15 Aug '15

[RESEND of the series after removing the extra patches] The SKL driver does not code DSP topology in driver. With this series the soc topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove CPU dai that is not used sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 173 ++-- sound/soc/intel/skylake/skl-topology.c | 1264 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1511 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 2.4.3

3 28

[alsa-devel] [asoc:topic/fsl-card 2/2] sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared
by kbuild test robot 15 Aug '15

15 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-card head: 50e0ee01382b8e08289d3db209738c5856fd25cf commit: 50e0ee01382b8e08289d3db209738c5856fd25cf [2/2] ASoC: fsl-asoc-card: add wm8960 support config: xtensa-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 50e0ee01382b8e08289d3db209738c5856fd25cf # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe': >> sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared (first use in this function) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; ^ sound/soc/fsl/fsl-asoc-card.c:486:29: note: each undeclared identifier is reported only once for each function it appears in vim +/WM8960_SYSCLK_AUTO +486 sound/soc/fsl/fsl-asoc-card.c 480 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; 481 priv->codec_priv.pll_id = WM8962_FLL; 482 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 483 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { 484 codec_dai_name = "wm8960-hifi"; 485 priv->card.set_bias_level = fsl_asoc_card_set_bias_level; > 486 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; 487 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; 488 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 489 } else { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: fsl: fix typos for sound/soc/fsl/*
by Xiubo Li 15 Aug '15

15 Aug '15

There are too much noise about the typos for fsl's drivers. So I fix all the typos here in this patch in almost every file I touched. Signed-off-by: Xiubo Li <lixiubo(a)cmss.chinamobile.com> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_sai.h | 12 ++++++------ sound/soc/fsl/fsl_spdif.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e1aa3834..883087f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -182,7 +182,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else { if (np) { - /* The eukrea,asoc-tlv320 driver was explicitely + /* The eukrea,asoc-tlv320 driver was explicitly * requested (through the device tree). */ dev_err(&pdev->dev, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..b4666fd 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -45,7 +45,7 @@ #define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) -/* SAI Transmit/Recieve Control Register */ +/* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) @@ -67,10 +67,10 @@ #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit and Recieve Configuration 1 Register */ +/* SAI Transmit and Receive Configuration 1 Register */ #define FSL_SAI_CR1_RFW_MASK 0x1f -/* SAI Transmit and Recieve Configuration 2 Register */ +/* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -82,12 +82,12 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) #define FSL_SAI_CR2_DIV_MASK 0xff -/* SAI Transmit and Recieve Configuration 3 Register */ +/* SAI Transmit and Receive Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -/* SAI Transmit and Recieve Configuration 4 Register */ +/* SAI Transmit and Receive Configuration 4 Register */ #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -97,7 +97,7 @@ #define FSL_SAI_CR4_FSP BIT(1) #define FSL_SAI_CR4_FSD_MSTR BIT(0) -/* SAI Transmit and Recieve Configuration 5 Register */ +/* SAI Transmit and Receive Configuration 5 Register */ #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8e93221..400d60e 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -707,7 +707,7 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, return ret; } -/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +/* Q-subcode information. The byte size is SPDIF_UBITS_SIZE/8 */ static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -739,7 +739,7 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit infomation */ +/* Valid bit information */ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -767,7 +767,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, return 0; } -/* DPLL lock infomation */ +/* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c0b940e..b759448 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -156,7 +156,7 @@ struct fsl_ssi_soc_data { * * @dbg_stats: Debugging statistics * - * @soc: SoC specifc data + * @soc: SoC specific data */ struct fsl_ssi_private { struct regmap *regs; @@ -1210,7 +1210,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, } } - /* For those SLAVE implementations, we ingore non-baudclk cases + /* For those SLAVE implementations, we ignore non-baudclk cases * and, instead, abandon MASTER mode that needs baud clock. */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config
by Axel Lin 15 Aug '15

15 Aug '15

Don't set .read_flag_mask for adav803, it's for adav801 only. Fixes: 0c2d69645628 ("ASoC: adav80x: Split SPI and I2C code into different modules") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 88fd37c..198c924 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -865,7 +865,6 @@ const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, - .read_flag_mask = 0x01, .max_register = ADAV80X_PLL_OUTE, -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: ssm4567: Add sense support
by Lars-Peter Clausen 15 Aug '15

15 Aug '15

The ssm4567 has sensing circuitry that can be used to monitor the current and voltage on the speaker amplifier output has well as the VBAT input. This data can be output over the I2S interface so it can be processed by a DSP or similar. This patch adds the sense capture output stream to the CODEC DAI as well as DAPM widgets that ensure that the sensing circuitry is powered up when the capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index ac242b2..e619d56 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -174,6 +174,12 @@ static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, &ssm4567_amplifier_boost_control), + SND_SOC_DAPM_SIGGEN("Sense"), + + SND_SOC_DAPM_PGA("Current Sense", SSM4567_REG_POWER_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Voltage Sense", SSM4567_REG_POWER_CTRL, 5, 1, NULL, 0), + SND_SOC_DAPM_PGA("VBAT Sense", SSM4567_REG_POWER_CTRL, 6, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -181,6 +187,13 @@ static const struct snd_soc_dapm_route ssm4567_routes[] = { { "OUT", NULL, "Amplifier Boost" }, { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, + + { "Current Sense", NULL, "Sense" }, + { "Voltage Sense", NULL, "Sense" }, + { "VBAT Sense", NULL, "Sense" }, + { "Capture Sense", NULL, "Current Sense" }, + { "Capture Sense", NULL, "Voltage Sense" }, + { "Capture Sense", NULL, "VBAT Sense" }, }; static int ssm4567_hw_params(struct snd_pcm_substream *substream, @@ -388,6 +401,14 @@ static struct snd_soc_dai_driver ssm4567_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32, }, + .capture = { + .stream_name = "Capture Sense", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, .ops = &ssm4567_dai_ops, }; -- 2.1.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: cs35l32: Use case range at appropriate places
by Axel Lin 15 Aug '15

15 Aug '15

Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. Below are the summary of readable/volatile/precious registers. The readable registers: 0x01 ~ 0x0D, 0x0F ~ 0x1C The volatile registers: 0x01 ~ 0x05, 0x15 ~ 0x18 The precious registers: 0x15 ~ 0x18 No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 45 +++++---------------------------------------- 1 file changed, 5 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 203c720..44c30fe 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -74,33 +74,8 @@ static const struct reg_default cs35l32_reg_defaults[] = { static bool cs35l32_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_PWRCTL1: - case CS35L32_PWRCTL2: - case CS35L32_CLK_CTL: - case CS35L32_BATT_THRESHOLD: - case CS35L32_VMON: - case CS35L32_BST_CPCP_CTL: - case CS35L32_IMON_SCALING: - case CS35L32_AUDIO_LED_MNGR: - case CS35L32_ADSP_CTL: - case CS35L32_CLASSD_CTL: - case CS35L32_PROTECT_CTL: - case CS35L32_INT_MASK_1: - case CS35L32_INT_MASK_2: - case CS35L32_INT_MASK_3: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: - case CS35L32_FLASH_MODE: - case CS35L32_MOVIE_MODE: - case CS35L32_FLASH_TIMER: - case CS35L32_FLASH_INHIBIT: + case CS35L32_DEVID_AB ... CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL ... CS35L32_FLASH_INHIBIT: return true; default: return false; @@ -110,15 +85,8 @@ static bool cs35l32_readable_register(struct device *dev, unsigned int reg) static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_DEVID_AB ... CS35L32_REV_ID: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -128,10 +96,7 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) static bool cs35l32_precious_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; -- 2.1.0

3 7

[alsa-devel] Applied "ASoC: cs42l56: Use case range at appropriate place" to the asoc tree
by Mark Brown 15 Aug '15

15 Aug '15

The patch ASoC: cs42l56: Use case range at appropriate place has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f325bfc933d140337e55507821d6bb021321059 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)ingics.com> Date: Wed, 12 Aug 2015 11:09:39 +0800 Subject: [PATCH] ASoC: cs42l56: Use case range at appropriate place The readable registers are in continuous range: 0x01 ~ 0x2e. Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 47 +--------------------------------------------- 1 file changed, 1 insertion(+), 46 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..98a68c6 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -115,52 +115,7 @@ static const struct reg_default cs42l56_reg_defaults[] = { static bool cs42l56_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L56_CHIP_ID_1: - case CS42L56_CHIP_ID_2: - case CS42L56_PWRCTL_1: - case CS42L56_PWRCTL_2: - case CS42L56_CLKCTL_1: - case CS42L56_CLKCTL_2: - case CS42L56_SERIAL_FMT: - case CS42L56_CLASSH_CTL: - case CS42L56_MISC_CTL: - case CS42L56_INT_STATUS: - case CS42L56_PLAYBACK_CTL: - case CS42L56_DSP_MUTE_CTL: - case CS42L56_ADCA_MIX_VOLUME: - case CS42L56_ADCB_MIX_VOLUME: - case CS42L56_PCMA_MIX_VOLUME: - case CS42L56_PCMB_MIX_VOLUME: - case CS42L56_ANAINPUT_ADV_VOLUME: - case CS42L56_DIGINPUT_ADV_VOLUME: - case CS42L56_MASTER_A_VOLUME: - case CS42L56_MASTER_B_VOLUME: - case CS42L56_BEEP_FREQ_ONTIME: - case CS42L56_BEEP_FREQ_OFFTIME: - case CS42L56_BEEP_TONE_CFG: - case CS42L56_TONE_CTL: - case CS42L56_CHAN_MIX_SWAP: - case CS42L56_AIN_REFCFG_ADC_MUX: - case CS42L56_HPF_CTL: - case CS42L56_MISC_ADC_CTL: - case CS42L56_GAIN_BIAS_CTL: - case CS42L56_PGAA_MUX_VOLUME: - case CS42L56_PGAB_MUX_VOLUME: - case CS42L56_ADCA_ATTENUATOR: - case CS42L56_ADCB_ATTENUATOR: - case CS42L56_ALC_EN_ATTACK_RATE: - case CS42L56_ALC_RELEASE_RATE: - case CS42L56_ALC_THRESHOLD: - case CS42L56_NOISE_GATE_CTL: - case CS42L56_ALC_LIM_SFT_ZC: - case CS42L56_AMUTE_HPLO_MUX: - case CS42L56_HPA_VOLUME: - case CS42L56_HPB_VOLUME: - case CS42L56_LOA_VOLUME: - case CS42L56_LOB_VOLUME: - case CS42L56_LIM_THRESHOLD_CTL: - case CS42L56_LIM_CTL_RELEASE_RATE: - case CS42L56_LIM_ATTACK_RATE: + case CS42L56_CHIP_ID_1 ... CS42L56_LIM_ATTACK_RATE: return true; default: return false; -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: fix runtime pm imbalance on error
by Junjie Mao 15 Aug '15

15 Aug '15

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Junjie Mao <junjie.mao(a)enight.me> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..27a9653aa3d9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -152,6 +152,7 @@ static int sst_power_control(struct device *dev, bool state) dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); -- 1.9.3

2 1

[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

2 1

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

2 1

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

3 3

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

2 1

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 1

[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8e5ff5e..37aabe3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

3 9

[alsa-devel] [RFC PATCH 08/10] ASoC: topology: change pass number of DAI smaller than graph
by mengdong.lin＠intel.com 15 Aug '15

15 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The PCM DAIs need to be loaded and added to ASoC core ealier than the graph (route). Otherwise, adding routes will fail for missing DAIs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1b6728b..0f8fa0d 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -45,12 +45,12 @@ #define SOC_TPLG_PASS_VENDOR 1 #define SOC_TPLG_PASS_MIXER 2 #define SOC_TPLG_PASS_WIDGET 3 -#define SOC_TPLG_PASS_GRAPH 4 -#define SOC_TPLG_PASS_PINS 5 -#define SOC_TPLG_PASS_PCM_DAI 6 +#define SOC_TPLG_PASS_PCM_DAI 4 +#define SOC_TPLG_PASS_GRAPH 5 +#define SOC_TPLG_PASS_PINS 6 #define SOC_TPLG_PASS_START SOC_TPLG_PASS_MANIFEST -#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PCM_DAI +#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PINS struct soc_tplg { const struct firmware *fw; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] USB audio fixes
by Pierre-Louis Bossart 14 Aug '15

14 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. Comments welcome. Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 14 Aug '15

14 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 111 ++++++++++++++++++++ 5 files changed, 135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..902daae --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,111 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int rate = params_rate(params); + + if (rate < ICS43432_RATE_MIN || rate > ICS43432_RATE_MAX) + return -EINVAL; + + return 0; +} + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .hw_params = ics43432_hw_params, + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Removed superfluous int32 checks - Check control value length in function - Compare against 0 - Use a tmp variable to access audio_feature_info[control-1] Thanks for review by Takashi Iwai :) --- sound/usb/mixer.c | 64 ++++++++++++++++++++++++++++++++++++++----------------- sound/usb/mixer.h | 2 ++ 2 files changed, 46 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..7046f54 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -282,6 +282,21 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) return val; } +static int uac2_ctl_value_size(int val_type) +{ + switch (val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + return 4; + case USB_MIXER_S16: + case USB_MIXER_U16: + return 2; + default: + return 1; + } + return 0; /* unreachable */ +} + /* * retrieve a mixer value @@ -328,14 +343,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + size = uac2_ctl_value_size(cval->val_type); } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +461,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + val_len = uac2_ctl_value_size(cval->val_type); /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +483,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +815,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1216,6 +1233,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; + struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; int nameid = uac_feature_unit_iFeature(desc); @@ -1241,7 +1259,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + ctl_info = &audio_feature_info[control-1]; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = ctl_info->type; + else /* UAC_VERSION_2 */ + cval->val_type = ctl_info->type_uac2 >= 0 ? + ctl_info->type_uac2 : ctl_info->type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 83 +++++++++++++++++++++++++++++++++++++++++-------------- sound/usb/mixer.h | 2 ++ 2 files changed, 65 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..675725b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -233,6 +233,14 @@ static int convert_signed_value(struct usb_mixer_elem_info *cval, int val) if (val >= 0x8000) val -= 0x10000; break; + case USB_MIXER_U32: + val &= 0xffffffff; + break; + case USB_MIXER_S32: + val &= 0xffffffff; + if (val >= 0x80000000) + val -= 0x100000000; + break; } return val; } @@ -253,6 +261,9 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) case USB_MIXER_S16: case USB_MIXER_U16: return val & 0xffff; + case USB_MIXER_S32: + case USB_MIXER_U32: + return val & 0xffffffff; } return 0; /* not reached */ } @@ -328,14 +339,26 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + size = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + size = 2; + break; + default: + size = 1; + break; + } } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +477,19 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + val_len = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + val_len = 2; + break; + default: + val_len = 1; + break; + } /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +503,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +835,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1241,7 +1278,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = audio_feature_info[control-1].type; + else /* UAC_VERSION_2 */ + cval->val_type = audio_feature_info[control-1].type_uac2 > -1 ? + audio_feature_info[control-1].type_uac2 : + audio_feature_info[control-1].type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 4

[alsa-devel] [PATCH v2 1/3] ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

Make sure to unlock the DAPM mutex when dapm_widget_list_create() fails. This means the function will now generate a trace_snd_soc_dapm_connected event, even if the creation of the list fails. But that was the behavior before the patch that introduced the unlock issue, so that should be fine. Fixes: 1ce43acff0c0 ("ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- New in v2 --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..c391bfe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,7 +1191,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, ret = dapm_widget_list_create(list, &widgets); if (ret) - return ret; + paths = ret; trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); -- 2.1.4

2 4

[alsa-devel] [PATCH 1/5] ASoC: dapm: Drop always true checks
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

list_first_entry() always returns non NULL and since the code previously checked that list is not empty it will also be a valid pointer. Furthermore a path has always a sink or a source widget. So both checks are redundant and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f834a0..10f7c7c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3485,11 +3485,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, list_source); - if (WARN_ON(!source_p || !sink_p) || - WARN_ON(!sink_p->source || !source_p->sink) || - WARN_ON(!source_p->source || !sink_p->sink)) - return -EINVAL; - source = source_p->source->priv; sink = sink_p->sink->priv; -- 2.1.4

2 11

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links
by Nicolin Chen 14 Aug '15

14 Aug '15

The dev->name of CODEC might not be identical to its codec_dai_name, so using dev->name to probe the CODEC dai is not a correct approach. This patch specifies each supporting codec_dai_name instead of using dev->name any more. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..040362f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -407,6 +407,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; struct clk *codec_clk; + const char *codec_dai_name; u32 width; int ret; @@ -459,6 +460,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name = "cs42888"; priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; @@ -467,9 +469,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + codec_dai_name = "wm8962"; priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; @@ -521,7 +525,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].codec_dai_name = codec_dai_name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +534,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; -- 1.9.1

2 1

[alsa-devel] [PATCH] Add rudimentary device tree support for SSM2518
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason the SSM2518 codec driver doesn't have DT support, even though there are bindings described for it in Documentation/devicetree/sound/ssm2518.txt . This patch adds rudimentary DT support for the compatible property which is a step forward. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] Add rudimentary DT support to SSM2518 codec driver
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason, the SSM2518 codec driver has device tree bindings specified (Documentation/devicetree/bindings/sound/ssm2518.txt) but the actual driver doesn't seem to support DT. This patch adds rudimentary support for the compatible property so that the driver can be used in a DT based setup. --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH 2/2] topology: Add API to set a vendor specific version number
by mengdong.lin＠intel.com 13 Aug '15

13 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This vendor-specific version number is optional. It will be written to the 'version' field of each block header of the binary toplogy data file. The vendor driver can check this number for further processing in kernel. The topology ABI version number is still stored in the 'abi' field of block headers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/topology.h b/include/topology.h index 6ff8c5f..9b84bd9 100644 --- a/include/topology.h +++ b/include/topology.h @@ -690,6 +690,14 @@ int snd_tplg_build(snd_tplg_t *tplg, const char *outfile); */ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len); +/** + * \brief Set an optional vendor specific version number. + * \param tplg Topology instance. + * \param version Vendor specific version number. + * \return Zero on success, otherwise a negative error code + */ +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version); + /* \} */ #ifdef __cplusplus diff --git a/src/topology/builder.c b/src/topology/builder.c index 3bccd44..2f01137 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -89,7 +89,7 @@ static int write_data_block(snd_tplg_t *tplg, int size, int tplg_type, /* write the header for this block */ ret = write_block_header(tplg, tplg_type, 0, - SND_SOC_TPLG_ABI_VERSION, 0, size, 1); + tplg->version, 0, size, 1); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); return ret; @@ -125,7 +125,7 @@ static int write_elem_block(snd_tplg_t *tplg, } ret = write_block_header(tplg, tplg_type, vendor_type, - SND_SOC_TPLG_ABI_VERSION, 0, size, count); + tplg->version, 0, size, count); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); @@ -243,7 +243,7 @@ static int write_manifest_data(snd_tplg_t *tplg) /* write the header for this block */ ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, - SND_SOC_TPLG_ABI_VERSION, 0, + tplg->version, 0, sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); if (ret < 0) { SNDERR("error: failed to write manifest block %d\n", ret); diff --git a/src/topology/parser.c b/src/topology/parser.c index 7ca244c..c0ab5e4 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -362,6 +362,13 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) return 0; } +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version) +{ + tplg->version = version; + + return 0; +} + void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) { tplg->verbose = verbose; -- 1.9.1

1 0

[alsa-devel] [PATCH 1/2] topology: Fix missing return value for snd_tplg_set_manifest_data()
by mengdong.lin＠intel.com 13 Aug '15

13 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/src/topology/parser.c b/src/topology/parser.c index ca7de06..7ca244c 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -358,6 +358,8 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) { tplg->manifest.priv.size = len; tplg->manifest_pdata = data; + + return 0; } void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix the white noise on Dell laptop
by woodrow.shen＠canonical.com 13 Aug '15

13 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Dell laptop causes the white noise by login screen and headphone, and the fixup function ALC292_FIXUP_DISABLE_AAMIX can eliminate this noise. Codec: Realtek ALC3235 Vendor Id: 0x10ec0293 Subsystem Id: 0x102806db Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1484334 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e3fe01..1de427a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5190,6 +5190,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.1.4

2 1

[alsa-devel] M-Audio transit: illegal sync type and noise on startup?
by Pierre-Louis Bossart 13 Aug '15

13 Aug '15

While playing with my M-Audio Transit USB device (0763:2006) I saw a synchronization type that doesn't exist in the USB audio class. I get audio out but often the start is garbled with white noise and cracks. Wondering what's going on. more /proc/asound/card1/stream0 M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous)u Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) <<<<<<< illegal Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) The only definitions in the spec are Asynchronous, Adaptive or Synchronous... lsusb tells me that there seems to be a synch endpoint for this setting: Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 100 Milliseconds Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 I couldn't figure out what the code does in this case. I didn't find any sync URBs being retired so wondering if this is considered as a synchronous or asynchronous endpoint based on the additional sync endpoint? Any idea on how to debug further? It looks like there could be a second error in the descriptors: if this is a synchronous endpoint then the sync endpoint is meaningless, and if it is asynchronous then the information on locking time is irrelevant... Regarding the noise on startup, should a delay be inserted after the mode is set? the spec recommends that the host sends silence periods for adaptive and synchronous modes. I see a set of mdelays being added in the quirks, should a new one be added? Thanks for any pointers, -Pierre

1 0

[alsa-devel] Questions about ASoC driver implementation
by Matt Campbell 12 Aug '15

12 Aug '15

Hi All, I'm diving into adding support for a new codec (AK4621) targeting IMX6 SoC and as this is my first dive into ALSA driver code (and first deep dive in to Linux driver development) I've come up with a few questions. I've done a lot of digging in available documentation as well as reading a lot of code, but some things remain unclear. First, I'm a little confused with what appears to be inconsistent terminology. The SoC overview.txt lists three types of component drivers: codec, platform and machine. However, in the code there are four types of snd_soc_register_* functions: card, platform, codec and component. From looking at the underlying structs and how these are used in existing code this is my best guess but please correct me if I'm wrong: codec -> snd_soc_register_codec machine -> snd_soc_register_card platform -> snd_soc_register_component and snd_soc_register_platform? (not sure about this, from what I can gather at some point there was an API switch to using component rather than platforms for the DAIs) My second question is around adding support for an amplifier external to the codec. The amplifier is a PGA controlled over SPI. One possible way I can see to adding this is to create a SPI based codec driver for the PGA that exposes an ALSA control for the gain level and give it a dummy DAI then add it to the card as an aux_dev. This feels a little kludgy as the PGA isn't a perfect fit for the codec class. Another option would be to have PGA driver separate from the ASoC framework, add the controls for the gain level directly in the ASoC card and put the glue code to interact with the device in the machine driver. This feels somewhat better to me, but has two problems. First, I don't know the best way to interact with the external PGA driver from the machine driver. Second, this leaves the PGA driver outside of the ASoC framework, If it was in ASoC it could be useful for others that come across this chip when writing machine drivers. Sorry for the long email, but there has been a bit of a learning curve up front. I tried to do my best with documentation and code reading before coming to the mailing list. I'll greatly appreciate any help the community can offer and look forward to contributing to the ALSA project. Best, ~Matt

2 2

[alsa-devel] [PATCH v2 1/4] drm/i915: Add audio set_ncts callback
by libin.yang＠intel.com 12 Aug '15

12 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the set_ncts callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..7305881 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,8 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*set_ncts)(struct device *, int port, int dev_entry, + int rate); } *ops; }; -- 1.9.1

5 35

[alsa-devel] [PATCH v3 0/3] topology: Add external API for building topology data
by Liam Girdwood 12 Aug '15

12 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Changes since V2:- o Fixed elem memory leaks on error handling paths o Fixed indentation. o Moved container_of and ARRAY_SIZE to local.h o remove use of unlink() and fixed file permissions. Changes since V1:- o Split out refactoring of OBJECT_TYPE to SND_TPLG_TYPE_ o Removed inclusing of type_compat.h from asoc.h Liam Girdwood (1): core: add convenience macros to local.h Mengdong Lin (2): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates include/local.h | 7 ++ include/topology.h | 202 ++++++++++++++++++++++++++++++++ src/topology/ctl.c | 292 ++++++++++++++++++++++++++++++++++++++++++++++ src/topology/dapm.c | 183 ++++++++++++++++++++++++++--- src/topology/elem.c | 17 +++ src/topology/parser.c | 59 +++++++++- src/topology/tplg_local.h | 26 ++--- 7 files changed, 749 insertions(+), 37 deletions(-) -- 2.1.4

2 4

[alsa-devel] [PATCH v3] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. There are no code changes required in the SAI driver since it has already wirten the word width to the corresponding register. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..d137290 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5640: fix line out no sound issue
by John Lin 11 Aug '15

11 Aug '15

The power for line out was not turned on when line out is enabled. So we add "LOUT amp" widget to turn on the power for line out. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5640.c | 40 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 37 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 7d488d8..3c5b496 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -985,6 +985,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, RT5640_PWR_LM); + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, 0); + break; + + default: + return 0; + } + + return 0; +} + static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1180,13 +1209,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, @@ -1501,8 +1533,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HP R Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HP L Playback"}, {"HPOR", NULL, "HP R Playback"}, - {"LOUTL", NULL, "LOUT MIX"}, - {"LOUTR", NULL, "LOUT MIX"}, + + {"LOUT amp", NULL, "LOUT MIX"}, + {"LOUTL", NULL, "LOUT amp"}, + {"LOUTR", NULL, "LOUT amp"}, }; static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v5] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 11 Aug '15

11 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 220 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 1ed0720..e3b7d0c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,110 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, + freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +725,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +740,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +776,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -721,11 +796,22 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, } } + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +866,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -831,9 +918,21 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, return ret; } } + + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +991,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1064,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1106,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1177,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1197,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Silence DMA slave ID compile warning on 64-bit
by Geert Uytterhoeven 11 Aug '15

11 Aug '15

On arm64: sound/soc/sh/rcar/dma.c: In function 'rsnd_dmaen_init': sound/soc/sh/rcar/dma.c:180:9: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] (void *)id); ^ include/linux/dmaengine.h:1185:75: note: in definition of macro 'dma_request_channel' #define dma_request_channel(mask, x, y) __dma_request_channel(&(mask), x, y) ^ Add an intermediate cast to "uintptr_t" to kill the compile warning. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 305b129298538850..bfbb8a5e93bdce6c 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -177,7 +177,7 @@ static int rsnd_dmaen_init(struct rsnd_dai_stream *io, dma_cap_set(DMA_SLAVE, mask); dmaen->chan = dma_request_channel(mask, shdma_chan_filter, - (void *)id); + (void *)(uintptr_t)id); } if (IS_ERR_OR_NULL(dmaen->chan)) { dmaen->chan = NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/3] topology: Add external API for building topology data
by Liam Girdwood 11 Aug '15

11 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Changes since V1:- o Split out refactoring of OBJECT_TYPE to SND_TPLG_TYPE_ o Removed inclusing of type_compat.h from asoc.h Liam Girdwood (1): topology: rename OBJECT_TYPE_ to SND_TPLG_TYPE_ Mengdong Lin (2): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates include/topology.h | 202 ++++++++++++++++++++++++++++ src/topology/builder.c | 38 +++--- src/topology/ctl.c | 329 +++++++++++++++++++++++++++++++++++++++++++--- src/topology/dapm.c | 223 ++++++++++++++++++++++++++----- src/topology/data.c | 10 +- src/topology/elem.c | 43 ++++-- src/topology/parser.c | 63 ++++++++- src/topology/pcm.c | 34 ++--- src/topology/text.c | 2 +- src/topology/tplg_local.h | 33 ++--- 10 files changed, 847 insertions(+), 130 deletions(-) -- 2.1.4

3 9

[alsa-devel] [RESEND PATCH] mfd: wm8994: Fix NULL pointer exception on missing pdata
by Inha Song 11 Aug '15

11 Aug '15

If we use the dev_get_platdata, pdata can be NULL in DT environments. Actually, NULL pointer exception is occurs with error messages like this: [ 1.226583] Unable to handle kernel NULL pointer dereference at virtual address 00000044 [ 1.467910] [<c02f1e38>] (wm8994_irq_init) from [<c02f1bd4>] (wm8994_i2c_probe+0x684/0x858) [ 1.476241] [<c02f1bd4>] (wm8994_i2c_probe) from [<c0371c78>] (i2c_device_probe+0x11c/0x17c) In wm8994-core, we already checked whether platform data is NULL and set the wm8994->pdata to platform data if platform data is not NULL. So, we can use wm8994->pdata instead of platform data. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm8994-irq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index 55c380a..55c59d9 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -193,7 +193,7 @@ int wm8994_irq_init(struct wm8994 *wm8994) { int ret; unsigned long irqflags; - struct wm8994_pdata *pdata = dev_get_platdata(wm8994->dev); + struct wm8994_pdata *pdata = &wm8994->pdata; if (!wm8994->irq) { dev_warn(wm8994->dev, -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [PATCH] topology: fix output file unlinking
by Liam Girdwood 11 Aug '15

11 Aug '15

Unlinking is unecessary as the old file will be overwritten with the new as intended. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/parser.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/src/topology/parser.c b/src/topology/parser.c index ed25bb8..5544517 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -266,9 +266,6 @@ int snd_tplg_build_file(snd_tplg_t *tplg, const char *infile, snd_config_t *cfg = NULL; int err = 0; - /* delete any old output files */ - unlink(outfile); - tplg->out_fd = open(outfile, O_RDWR | O_CREAT, S_IRWXU | S_IRWXG | S_IRWXO); if (tplg->out_fd < 0) { -- 2.1.4

2 2

[alsa-devel] [PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk
by Takashi Sakamoto 11 Aug '15

11 Aug '15

Some users have reported current ALSA Fireworks driver stops packet streaming due to detecting discontinuity. [alsa-devel] ALSA snd-fireworks Echo AudioFire 2 http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [alsa-devel] AudioFire4 quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095697.html Linux 3.16 or later have this issue. As long as I investigate, this issue depends on firmware version installed in these devices. While, it's not clear that which version of firmwares for which model has this quirk. At least, firmware version 5.7.0, 5.7.3 and 5.8.0 for AudioFire2/AudioFire4/AudioFirePre8 have this quirk. This patchset applies a solution for this issue, with my assumption that 'the same firmware binary has the same quirk'. These models, AudioFire8 (since Jul 2009) and Gibson Robot Interface Pack (RIP) series uses the same firmware binary in driver package. If possible, the second patch should be applied to all of stable kernels. This patchset updates my previous RFC: [alsa-devel] [RFC][PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/095849.html Takashi Sakamoto (2): Revert "ALSA: fireworks: add support for AudioFire2 quirk" ALSA: fireworks/firewire-lib: add support for recent firmware quirk sound/firewire/amdtp.c | 5 +++-- sound/firewire/amdtp.h | 2 ++ sound/firewire/fireworks/fireworks.c | 10 ++++++++-- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_stream.c | 12 +++++++++--- 5 files changed, 23 insertions(+), 8 deletions(-) -- 2.1.4

2 7

[alsa-devel] [PATCH v4] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 11 Aug '15

11 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 220 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 1ed0720..e7ee49a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,110 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, + freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +725,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +740,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +776,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -721,11 +796,22 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, } } + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +866,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -831,9 +918,21 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, return ret; } } + + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +991,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1064,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1106,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1177,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1197,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..40280ce 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -1,5 +1,5 @@ /* - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * 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 @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

3 8

[alsa-devel] [PATCH] jack: Fixing a small memory leak
by Valentin Corfu 11 Aug '15

11 Aug '15

A small memory leak can be observed when using alsa-plugins (jack). Check the following valgrind log: ... ==1034== 8 bytes in 1 blocks are definitely lost in loss record 7 of 108 ==1034== at 0x4027E5A: calloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==1034== by 0x4035102: snd_pcm_jack_prepare (in /usr/lib/alsa-lib/libasound_module_pcm_jack.so) ==1034== by 0x40D07AB: snd_pcm_ioplug_prepare (pcm_ioplug.c:145) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x4084A1D: snd_pcm_hw_params (pcm.c:838) ==1034== by 0x4090013: snd_pcm_set_params (pcm.c:8053) ==1034== by 0x80486EB: main (in /root/pcm_min_ok) ... ==1034== LEAK SUMMARY: ==1034== definitely lost: 8 bytes in 1 blocks ==1034== indirectly lost: 0 bytes in 0 blocks ==1034== possibly lost: 31,311 bytes in 1,630 blocks ==1034== still reachable: 5,619 bytes in 24 blocks ==1034== suppressed: 0 bytes in 0 blocks ... Signed-off-by: Valentin Corfu <corfuvalentin(a)gmail.com> --- jack/pcm_jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 568ec43..de9c46e 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -103,6 +103,7 @@ static void snd_pcm_jack_free(snd_pcm_jack_t *jack) if (jack->io.poll_fd >= 0) close(jack->io.poll_fd); free(jack->areas); + free(jack->ports); free(jack); } } -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 11 Aug '15

11 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 68 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 944 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 276 ++++++ 5 files changed, 1294 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..46e1868 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,68 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-pullup: Pull-up JKDET pin. Boolean. + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30ms + 1 - 50ms + 2 - 100ms + 3 - 30ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-pullup; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedane according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0x14 0x29 0x40 0x78>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(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 diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..96eed3a --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,944 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; + int micbias_voltage; + bool jkdet_pullup; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 3", + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 3", + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"HPOL", NULL, "Output Driver L Stage 3"}, + {"HPOR", NULL, "Output Driver R Stage 3"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, 0x0400, 0x0400); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_SELECT, + nau8825->jkdet_pullup ? NAU8825_JKDET_PULL_SELECT : 0); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + /* Unmask needed IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, + unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_pullup = device_property_read_bool(dev, + "nuvoton,jkdet-pullup"); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + * Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..bb103f0 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,276 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * 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 __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +/* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_SELECT (1 << 11) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - Normal mode, 1 - Output Hi-Z */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0

1 0

[alsa-devel] [PATCH 1/4] drm/i915: Add audio set_ncts callback
by libin.yang＠intel.com 11 Aug '15

11 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the set_ncts callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..7305881 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,8 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*set_ncts)(struct device *, int port, int dev_entry, + int rate); } *ops; }; -- 1.9.1

4 11

[alsa-devel] [PATCH v2] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. There are no code changes required in the SAI driver since it has already wirten the word width to the corresponding register. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..40280ce 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -1,5 +1,5 @@ /* - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * 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 @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

2 2

[alsa-devel] jack: Fixing a small memory leak
by Valentin Corfu 10 Aug '15

10 Aug '15

Hello Alsa developers, I observed a small memory leak in alsa-plugins (jack): ... ==1034== ==1034== 8 bytes in 1 blocks are definitely lost in loss record 7 of 108 ==1034== at 0x4027E5A: calloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==1034== by 0x4035102: snd_pcm_jack_prepare (in /usr/lib/alsa-lib/libasound_module_pcm_jack.so) ==1034== by 0x40D07AB: snd_pcm_ioplug_prepare (pcm_ioplug.c:145) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x4084A1D: snd_pcm_hw_params (pcm.c:838) ==1034== by 0x4090013: snd_pcm_set_params (pcm.c:8053) ==1034== by 0x80486EB: main (in /root/pcm_min_ok) ... ==1034== LEAK SUMMARY: ==1034== definitely lost: 8 bytes in 1 blocks ==1034== indirectly lost: 0 bytes in 0 blocks ==1034== possibly lost: 31,311 bytes in 1,630 blocks ==1034== still reachable: 5,619 bytes in 24 blocks ==1034== suppressed: 0 bytes in 0 blocks ==1034== Reachable blocks (those to which a pointer was found) are not shown. ==1034== To see them, rerun with: --leak-check=full --show-leak-kinds=all ... Best Regards, Valentin Corfu

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[alsa-devel] [PATCH] ALSA: hda - Fix pin config and mapping on Alienware 15
by Takashi Iwai 10 Aug '15

10 Aug '15

Alienware 15 has CA0132 codec for its onboard sound, but the pin config and mapping seem quite different from other Creative boards. This patch corrects them, at least, for providing the right headphone and mic jack notification, as well as removing the non-existing SPDIF pins. Even with this fix, not all stuff works perfectly yet, mainly because of the badly written ca0132 driver code -- it has too many implicit assumptions of pin configs and maps. Nevertheless, this is a small good step forward. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101981 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 87 ++++++++++++++++++++++++++++++-------------- 1 file changed, 59 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0f039abe9673..186792fe226e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -763,6 +763,20 @@ enum { QUIRK_ALIENWARE, }; +static const struct hda_pintbl alienware_pincfgs[] = { + { 0x0b, 0x90170110 }, /* Builtin Speaker */ + { 0x0c, 0x411111f0 }, /* N/A */ + { 0x0d, 0x411111f0 }, /* N/A */ + { 0x0e, 0x411111f0 }, /* N/A */ + { 0x0f, 0x0321101f }, /* HP */ + { 0x10, 0x411111f0 }, /* Headset? disabled for now */ + { 0x11, 0x03a11021 }, /* Mic */ + { 0x12, 0xd5a30140 }, /* Builtin Mic */ + { 0x13, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE), {} @@ -3147,7 +3161,7 @@ static int ca0132_select_out(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->out_pins[1]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp); else jack_present = spec->vnode_lswitch[VNID_HP_SEL - VNODE_START_NID]; @@ -3309,7 +3323,7 @@ static int ca0132_select_mic(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->input_pins[0]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_amic1); else jack_present = spec->vnode_lswitch[VNID_AMIC1_SEL - VNODE_START_NID]; @@ -4617,37 +4631,54 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.num_dacs = 3; spec->multiout.max_channels = 2; - spec->num_outputs = 2; - spec->out_pins[0] = 0x0b; /* speaker out */ if (spec->quirk == QUIRK_ALIENWARE) { codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); + snd_hda_apply_pincfgs(codec, alienware_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x0f; - } else{ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = 0x0f; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = 0x11; + } else { + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } - spec->shared_out_nid = 0x2; - spec->unsol_tag_hp = spec->out_pins[1]; - - spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ - spec->adcs[1] = 0x8; /* analog mic2 */ - spec->adcs[2] = 0xa; /* what u hear */ - - spec->num_inputs = 3; - spec->input_pins[0] = 0x12; - spec->input_pins[1] = 0x11; - spec->input_pins[2] = 0x13; - spec->shared_mic_nid = 0x7; - spec->unsol_tag_amic1 = spec->input_pins[0]; - - /* SPDIF I/O */ - spec->dig_out = 0x05; - spec->multiout.dig_out_nid = spec->dig_out; - cfg->dig_out_pins[0] = 0x0c; - cfg->dig_outs = 1; - cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; - spec->dig_in = 0x09; - cfg->dig_in_pin = 0x0e; - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static int ca0132_prepare_verbs(struct hda_codec *codec) -- 2.5.0

1 0

[alsa-devel] [PATCH] ALSA: echoaudio: Use standard C definitions of true and false
by Mark Brown 10 Aug '15

10 Aug '15

The echoaudio locally defines TRUE and FALSE. Not only is this redundant given that C now has a boolean type it results in lots of warnings as other headers also define these macros, causing duplicate definitions. Fix this by removing the local defines and converting all local users to use the standard C true and false instead, simply removing the macros is less safe due to implicit inclusion of the other definitons. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/pci/echoaudio/darla20_dsp.c | 6 +++--- sound/pci/echoaudio/darla24_dsp.c | 6 +++--- sound/pci/echoaudio/echo3g_dsp.c | 16 ++++++++-------- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/echoaudio/echoaudio.h | 7 ++----- sound/pci/echoaudio/echoaudio_3g.c | 12 ++++++------ sound/pci/echoaudio/echoaudio_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/gina20_dsp.c | 8 ++++---- sound/pci/echoaudio/gina24_dsp.c | 22 +++++++++++----------- sound/pci/echoaudio/indigo_dsp.c | 6 +++--- sound/pci/echoaudio/indigodj_dsp.c | 6 +++--- sound/pci/echoaudio/indigodjx_dsp.c | 6 +++--- sound/pci/echoaudio/indigoio_dsp.c | 6 +++--- sound/pci/echoaudio/indigoiox_dsp.c | 6 +++--- sound/pci/echoaudio/layla20_dsp.c | 12 ++++++------ sound/pci/echoaudio/layla24_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/mia_dsp.c | 8 ++++---- sound/pci/echoaudio/mona_dsp.c | 20 ++++++++++---------- 18 files changed, 99 insertions(+), 102 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index febee5b..320837b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -44,18 +44,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 7b4f6fd..8736b5e 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -44,17 +44,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_ESYNC; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index ae11ce1..6deb80c 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -59,8 +59,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get @@ -78,8 +78,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->px_analog_in = chip->bx_analog_in = 14; chip->px_digital_in = chip->bx_digital_in = 16; chip->px_num = chip->bx_num = 24; - chip->has_phantom_power = TRUE; - chip->hasnt_input_nominal_level = TRUE; + chip->has_phantom_power = true; + chip->hasnt_input_nominal_level = true; } else if (err == E3G_LAYLA3G_BOX_TYPE) { chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -106,10 +106,10 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - chip->phantom_power = FALSE; + chip->professional_spdif = false; + chip->non_audio_spdif = false; + chip->bad_board = false; + chip->phantom_power = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 862db9a..1cb85ae 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2245,7 +2245,7 @@ static int snd_echo_resume(struct device *dev) #ifdef ECHOCARD_HAS_MIDI if (chip->midi_input_enabled) - enable_midi_input(chip, TRUE); + enable_midi_input(chip, true); if (chip->midi_out) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 3251522..152ce15 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -153,9 +153,6 @@ #define _ECHOAUDIO_H_ -#define TRUE 1 -#define FALSE 0 - #include "echoaudio_dsp.h" @@ -378,8 +375,8 @@ struct echoaudio { */ u8 output_clock; /* Layla20 only */ char meters_enabled; /* VU-meters status */ - char asic_loaded; /* Set TRUE when ASIC loaded */ - char bad_board; /* Set TRUE if DSP won't load */ + char asic_loaded; /* Set true when ASIC loaded */ + char bad_board; /* Set true if DSP won't load */ char professional_spdif; /* 0 = consumer; 1 = professional */ char non_audio_spdif; /* 3G - only */ char digital_in_automute; /* Gina24, Layla24, Mona - only */ diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 2fa66dc..22c786b 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -41,7 +41,7 @@ static int check_asic_status(struct echoaudio *chip) return -EIO; chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); - chip->asic_loaded = FALSE; + chip->asic_loaded = false; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); @@ -55,7 +55,7 @@ static int check_asic_status(struct echoaudio *chip) if (box_status == E3G_ASIC_NOT_LOADED) return -ENODEV; - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return box_status & E3G_BOX_TYPE_MASK; } @@ -243,7 +243,7 @@ static int load_asic(struct echoaudio *chip) * 48 kHz, internal clock, S/PDIF RCA mode */ if (box_type >= 0) { err = write_control_reg(chip, E3G_48KHZ, - E3G_FREQ_REG_DEFAULT, TRUE); + E3G_FREQ_REG_DEFAULT, true); if (err < 0) return err; } @@ -378,16 +378,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 1a9427a..15aae2f 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -103,8 +103,8 @@ static int write_dsp(struct echoaudio *chip, u32 data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_dbg(chip->card->dev, "write_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_dbg(chip->card->dev, "write_dsp: Set bad_board to true\n"); return -EIO; } @@ -126,8 +126,8 @@ static int read_dsp(struct echoaudio *chip, u32 *data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_err(chip->card->dev, "read_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_err(chip->card->dev, "read_dsp: Set bad_board to true\n"); return -EIO; } @@ -166,7 +166,7 @@ static int read_sn(struct echoaudio *chip) /* This card has no ASIC, just return ok */ static inline int check_asic_status(struct echoaudio *chip) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } @@ -341,11 +341,11 @@ static int load_dsp(struct echoaudio *chip, u16 *code) dev_warn(chip->card->dev, "DSP is already loaded!\n"); return 0; } - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */ + chip->asic_loaded = false; /* Loading the DSP code will reset the ASIC */ - dev_dbg(chip->card->dev, "load_dsp: Set bad_board to TRUE\n"); + dev_dbg(chip->card->dev, "load_dsp: Set bad_board to true\n"); /* If this board requires a resident loader, install it. */ #ifdef DSP_56361 @@ -471,7 +471,7 @@ static int load_dsp(struct echoaudio *chip, u16 *code) } chip->dsp_code = code; /* Show which DSP code loaded */ - chip->bad_board = FALSE; /* DSP OK */ + chip->bad_board = false; /* DSP OK */ return 0; } udelay(100); @@ -951,10 +951,10 @@ static int rest_in_peace(struct echoaudio *chip) /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); - set_meters_on(chip, FALSE); + set_meters_on(chip, false); #ifdef ECHOCARD_HAS_MIDI - enable_midi_input(chip, FALSE); + enable_midi_input(chip, false); #endif /* Go to sleep */ @@ -981,9 +981,9 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init all the basic stuff */ chip->card_name = ECHOCARD_NAME; - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; + chip->asic_loaded = false; memset(chip->comm_page, 0, sizeof(struct comm_page)); /* Init the comm page */ diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 5dafe92..b237757 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -48,19 +48,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 6971766..8eff2b4 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | @@ -76,7 +76,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -86,8 +86,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -152,7 +152,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; } @@ -226,7 +226,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->sample_rate = rate; dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, clock); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -274,7 +274,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -285,17 +285,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_CDROM: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -333,7 +333,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 54edd67..c97dc83 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 2cf5cc0..2428b35 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 5252863..5fbd4a3 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 4e81787..79b68ba 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 6de3f9b..1ae394e 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index f2024a3..5e5b6e2 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -62,7 +62,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -71,7 +71,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } @@ -113,7 +113,7 @@ static int check_asic_status(struct echoaudio *chip) u32 asic_status; int goodcnt, i; - chip->asic_loaded = FALSE; + chip->asic_loaded = false; for (i = goodcnt = 0; i < 5; i++) { send_vector(chip, DSP_VC_TEST_ASIC); @@ -127,7 +127,7 @@ static int check_asic_status(struct echoaudio *chip) if (asic_status == ASIC_ALREADY_LOADED) { if (++goodcnt == 3) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } } diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 4f11e81..df28e51 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -64,7 +64,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; if ((err = init_line_levels(chip)) < 0) return err; @@ -77,8 +77,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -135,7 +135,7 @@ static int load_asic(struct echoaudio *chip) err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, FW_LAYLA24_2S_ASIC); if (err < 0) - return FALSE; + return false; /* Now give the external ASIC a little time to set up */ mdelay(10); @@ -147,7 +147,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, - TRUE); + true); return err; } @@ -241,7 +241,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, control_reg); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -287,7 +287,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -334,17 +334,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) short asic; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2A_ASIC; break; default: @@ -383,7 +383,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index fdad079..8f612a0 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -52,19 +52,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; if ((subdevice_id & 0x0000f) == MIA_MIDI_REV) - chip->has_midi = TRUE; + chip->has_midi = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index 843c7a5..dce9e57 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -79,8 +79,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -148,7 +148,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; @@ -356,7 +356,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -367,16 +367,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -415,7 +415,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, FALSE); + err = write_control_reg(chip, control_reg, false); spin_unlock_irq(&chip->lock); if (err < 0) return err; -- 2.5.0

2 2

[alsa-devel] [RFC PATCH 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch set allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: soc_bind_dai_link() change 2nd argument to DAI link pointer ASoC: implement DAI links in a list. ASoC: Add support for dummy DAI links and PCM runtimes ASoC: bind new DAI links after probing components. ASoC: support adding a DAI dynamically ASoC: topology: change pass number of DAI smaller than graph ASoC: Topology: change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

3 2

[alsa-devel] writing an alsa driver
by Johnny 10 Aug '15

10 Aug '15

Hi, I have been following your progress with great anticipation. > SSP2 > http://permalink.gmane.org/gmane.linux.alsa.devel/140328 > future versions of the firmware should support SSP1 So, I'm guessing, until the firmware supports SSP1, there will not be any sound? I also tried using the firmware that supports SSP0, which Vinod posted in another thread, and get no sound. Regards, John

1 0

[alsa-devel] [PATCH v3] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 10 Aug '15

10 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 217 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 177 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..e9f8e41 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,109 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +724,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +739,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +775,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -711,6 +785,9 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: switch (snd_soc_codec_get_bias_level(codec)) { case SND_SOC_BIAS_STANDBY: + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -726,6 +803,13 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +864,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -822,6 +907,10 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, msleep(100); + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -834,6 +923,13 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +988,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1061,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1103,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1174,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1194,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 5

[alsa-devel] Support for snd_pcm_link() on the rate plugin
by Christian Gruber 10 Aug '15

10 Aug '15

Dear ALSA community, I have a question regarding the usage of the function snd_pcm_link() on the rate plugin. As can be seen from its implementation (see pcm.c, line 1368), the function snd_pcm_link() calls the fast_op "link" of the first pcm device/plugin to be linked, which is a function pointer, whose destination depends on the type of pcm device/plugin to be linked. If the device is a HW-device, the function snd_pcm_hw_link() is called for instance (see pcm_hw.c). Several other plugins use the generic function snd_pcm_generic_link() (see pcm_generic.c, line 212), which forwards the function call to their slave devices/plugins, e.g. the plugins mmap_emul, linear, lfloat, mulaw, alaw, route, etc. The rate plugin instead does not support snd_pcm_link(), since the function pointer "link" is not assigned in the definition of snd_pcm_rate_fast_ops (see pcm_rate.c, line 1153). Why does the rate plugin not support snd_pcm_link()? Is their any reason to not use the generic function snd_pcm_generic_link() within the rate plugin? If yes, what's the reason for it? Thanks in advance for answering this question, Christian

1 0

[alsa-devel] [PATCH 1/2] ASoC: extend seq_notifier callback interface
by yang.a.fang＠intel.com 10 Aug '15

10 Aug '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This is needed for some codec driver to know whether it is power up/down sequence call from dapm Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/soc-dapm.c | 5 ++--- 6 files changed, 7 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 37d95a8..1d017a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -607,7 +607,7 @@ struct snd_soc_dapm_context { /* Go to BIAS_OFF in suspend if the DAPM context is idle */ unsigned int suspend_bias_off:1; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ diff --git a/include/sound/soc.h b/include/sound/soc.h index add5097..0072e21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -875,7 +875,7 @@ struct snd_soc_codec_driver { bool suspend_bias_off; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8120b0c..6f4e6c3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -733,7 +733,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), }; static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b011253..12da3b1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -280,7 +280,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, #define WM8903_DCS_MODE_START_STOP 2 static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a696db..f12a900 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -686,7 +686,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1992568..ebea2d3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1528,8 +1528,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, - cur_subseq); + i, cur_subseq, power_up); } if (cur_dapm && w->dapm != cur_dapm) @@ -1591,7 +1590,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, cur_subseq); + i, cur_subseq, power_up); } list_for_each_entry(d, &card->dapm_list, list) { -- 1.7.9.5

4 7

[alsa-devel] [PATCH 00/12] Add ASoC support for AMD APUs
by Alex Deucher 10 Aug '15

10 Aug '15

Hi, This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create an amd gnb (graphics northbridge) bus which we can hang the i2s codec on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the gnb bus and the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. Patch 3 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream Thanks, Alex Chunming Zhou (2): drm/amdgpu: add amd_gnb_bus support drm/amd: add ACP 2.x register headers Maruthi Bayyavarapu (1): drm/amd: add ACP driver support (v4) Maruthi Srinivas Bayyavarapu (9): drm/amd: remove amd gnb bus default runtime pm ops drm/amd: modify ACP DMA buffer position update logic (v2) drm/amd: add ACP suspend/resume functionality drm/amd: change ACP SRAM banks used for audio drm/amd: shutdown unused ACP SRAM memory banks drm/amd: remove bitfield usage in ACP IP module ASoC: AMD: add AMD ASoC ACP-I2S driver (v2) ASoC: AMD: add suspend/resume for ACP PCM driver ASoC: AMD: add ACP PCM driver runtime PM drivers/gpu/drm/Kconfig | 5 + drivers/gpu/drm/Makefile | 1 + drivers/gpu/drm/amd/acp/Kconfig | 9 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 1134 ++++++++++ drivers/gpu/drm/amd/acp/acp_hw.h | 91 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/acp/include/amd_acp.h | 214 ++ drivers/gpu/drm/amd/amdgpu/Makefile | 14 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 11 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 201 ++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 40 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/bus/Kconfig | 7 + drivers/gpu/drm/amd/bus/Makefile | 4 + drivers/gpu/drm/amd/bus/amd_gnb_bus.c | 261 +++ drivers/gpu/drm/amd/include/amd_shared.h | 1 + .../gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h | 437 ++++ .../drm/amd/include/asic_reg/acp/acp_2_1_enum.h | 1198 ++++++++++ .../drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h | 1568 +++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h | 609 ++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_enum.h | 1068 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h | 582 +++++ .../drm/amd/include/asic_reg/acp/acp_2_3_enum.h | 1079 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h | 2192 +++++++++++++++++++ drivers/gpu/drm/amd/include/bus/amd_gnb_bus.h | 78 + sound/soc/Kconfig | 2 +- sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 13 + sound/soc/amd/Makefile | 11 + sound/soc/amd/acp-pcm-dma.c | 759 +++++++ sound/soc/amd/acp-rt286.c | 264 +++ 33 files changed, 14214 insertions(+), 2 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.h create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/acp/include/amd_acp.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 drivers/gpu/drm/amd/bus/Kconfig create mode 100644 drivers/gpu/drm/amd/bus/Makefile create mode 100644 drivers/gpu/drm/amd/bus/amd_gnb_bus.c create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/bus/amd_gnb_bus.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp-rt286.c -- 1.8.3.1

5 24

[alsa-devel] [PATCH 0/3] topology: Add external API for building topology data.
by Liam Girdwood 10 Aug '15

10 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Mengdong Lin (3): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates topology: add support for adding bytes controls to widgets include/sound/asoc.h | 2 + include/topology.h | 202 ++++++++++++++++++++++++++++ src/topology/builder.c | 38 +++--- src/topology/ctl.c | 329 +++++++++++++++++++++++++++++++++++++++++++--- src/topology/dapm.c | 226 ++++++++++++++++++++++++++----- src/topology/data.c | 10 +- src/topology/elem.c | 43 ++++-- src/topology/parser.c | 63 ++++++++- src/topology/pcm.c | 34 ++--- src/topology/text.c | 2 +- src/topology/tplg_local.h | 33 ++--- 11 files changed, 849 insertions(+), 133 deletions(-) -- 2.1.4

2 5

[alsa-devel] ALSA: es1968: Use monotonic clock
by Abhilash Jindal 10 Aug '15

10 Aug '15

Wall time obtained from do_gettimeofday is susceptible to sudden jumps due to user setting the time or due to NTP. Monotonic time is constantly increasing time better suited for comparing two timestamps. --- sound/pci/es1968.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index b0e3d92..6830c31 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1710,7 +1710,7 @@ static void es1968_measure_clock(struct es1968 *chip) int i, apu; unsigned int pa, offset, t; struct esm_memory *memory; - struct timeval start_time, stop_time; + struct timespec start_time, stop_time; if (chip->clock == 0) chip->clock = 48000; /* default clock value */ @@ -1759,12 +1759,12 @@ static void es1968_measure_clock(struct es1968 *chip) snd_es1968_bob_inc(chip, ESM_BOB_FREQ); __apu_set_register(chip, apu, 5, pa & 0xffff); snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); - do_gettimeofday(&start_time); + ktime_get_ts(&start_time); spin_unlock_irq(&chip->reg_lock); msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); - do_gettimeofday(&stop_time); + ktime_get_ts(&stop_time); snd_es1968_trigger_apu(chip, apu, 0); /* stop */ snd_es1968_bob_dec(chip); chip->in_measurement = 0; @@ -1776,11 +1776,11 @@ static void es1968_measure_clock(struct es1968 *chip) offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2); t = stop_time.tv_sec - start_time.tv_sec; - t *= 1000000; - if (stop_time.tv_usec < start_time.tv_usec) - t -= start_time.tv_usec - stop_time.tv_usec; + t *= USEC_PER_SEC; + if (stop_time.tv_nsec < start_time.tv_nsec) + t -= (start_time.tv_nsec - stop_time.tv_nsec)/NSEC_PER_USEC; else - t += stop_time.tv_usec - start_time.tv_usec; + t += (stop_time.tv_nsec - start_time.tv_nsec)/NSEC_PER_USEC; if (t == 0) { snd_printk(KERN_ERR "?? calculation error..\n"); } else { -- 1.7.9.5

2 1

[alsa-devel] RME MADIFX mainline inclusion
by Adrian Knoth 10 Aug '15

10 Aug '15

Hi Takashi! Back in the days (late 2012 I guess), I wrote the MADIFX driver based on the existing RME hdspm.c. I never had physical access to this 1399 Euro audio interface, only the OSX driver and a shell login to one of IOSONO's machines which temporarily featured such a card. I know there are other users like University of Washington (DXARTS) and probably some more (the usual WFS suspects). Long story short: I don't have such a card and since I no longer work in Linux audio, I guess I won't have access in the foreseeable future. That said, I think the driver should be part of the mainline kernel, so it gets all the love the other drivers get (static code analysis, API changes etc.) I think I've ported everything you guys have done to hdspm: I've read the commitlog for hdspm and applied those changes to the madifx driver. Code is here: https://github.com/adiknoth/madifx/commits/tokernel Status: * PCM playback/capture working (SS and DS tested, QS untested) * MIDI working * All card settings working (e.g. TX64, SMUX, AESpro, WC-Term, WC-singlespeed...) * Slave Mode/External clock selection working * Mirror-MADI1-to-Out2+3 maybe working (untested) * Redundancy mode maybe working (untested) * ioctls implemented (see ioctl.c) * Static mixer working (fixed 1:1 mapping) * DSP NOT working. RME doesn't intend to release any information regarding the DSP. * Adjustable mixer NOT working (needs new userspace tools) * Levelmetering NOT working (maybe wrong, needs new userspace tools) I have no idea who could write the new mixer code. Apparently, demand is low, maybe zero. checkpatch reports four warnings of which one is a false-positive, the other one doesn't make sense and only two are real but somewhat ugly to fix (line over 80 with really long array indices). In its current state, can I send you three patches (Kconfig, madifx.h, madifx.c) that add said driver to sound/pci/rme9652? Cheers

6 16

[alsa-devel] DP MST audio support
by Dave Airlie 10 Aug '15

10 Aug '15

Hi, So I have a branch that makes no sound, http://cgit.freedesktop.org/~airlied/linux/log/?h=dp-mst-audio and I'm not sure where I need to turn to next, The Intel docs I've read are kinda vague, assuming you know lots of things I clearly don't. so in theory my branch, sets up the SDP stream to the monitor in the payload creation, enables the codec in the intel GPU driver, and passes the ELD to the audio driver. The audio driver uses the device list to get the presence/valid bits per device, and manages to retrieve the ELD. I even create ELD files in /proc/asound/HDMI/ that have sensible values in them So it looks like I'm just missing some routing somewhere, most likely in the audio driver, then again I could be missing a lot more than that. Just looking for any ideas or knowledge people may have locked in their brains or inside their firewalls. Dave.

4 6

[alsa-devel] [RFC PATCH 10/10] ASOC: topology: Add PCM DAIs dynamically when loading them
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Add PCM DAIs to ASoC core when loading them from topology. Embed dobj in the DAI to free it when unloading. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..e8f0404 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -207,6 +207,7 @@ struct snd_soc_dai_driver { const char *name; unsigned int id; unsigned int base; + struct snd_soc_dobj dobj; /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 5405f64..a49c134 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -71,7 +71,6 @@ struct snd_soc_dobj { union { struct snd_soc_dobj_control control; struct snd_soc_dobj_widget widget; - struct snd_soc_dobj_pcm_dai pcm_dai; }; void *private; /* core does not touch this */ }; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0f8fa0d..6c0dd3f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -499,6 +499,9 @@ static void remove_widget(struct snd_soc_component *comp, static void remove_pcm_dai(struct snd_soc_component *comp, struct snd_soc_dobj *dobj, int pass) { + struct snd_soc_dai_driver *dai = + container_of(dobj, struct snd_soc_dai_driver, dobj); + if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -506,7 +509,7 @@ static void remove_pcm_dai(struct snd_soc_component *comp, dobj->ops->pcm_dai_unload(comp, dobj); list_del(&dobj->list); - kfree(dobj); + kfree(dai); } /* bind a kcontrol to it's IO handlers */ @@ -1558,12 +1561,75 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } +static unsigned int get_rates(unsigned int rate_min, unsigned int rate_max) +{ + static int pcm_rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, + 64000, 88200, 96000, 176400, 192000, + }; + unsigned int rates = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(pcm_rates); i++) { + if (pcm_rates[i] >= rate_min && pcm_rates[i] <= rate_max) + rates |= 1 << i; + } + + return rates; +} + +static int soc_tplg_pcm_dai_create(struct soc_tplg *tplg, + struct snd_soc_tplg_pcm_dai *dai) +{ + struct snd_soc_dai_driver *dai_drv; + struct snd_soc_pcm_stream *stream; + struct snd_soc_tplg_stream_caps *caps; + + dai_drv = kzalloc(sizeof(struct snd_soc_dai_driver), GFP_KERNEL); + if (dai_drv == NULL) + return -ENOMEM; + + dai_drv->name = dai->name; + dai_drv->id = dai->id; + dai_drv->compress_dai = dai->compress; + + if (dai->playback) { + stream = &dai_drv->playback; + caps = &dai->capconf[SND_SOC_TPLG_STREAM_PLAYBACK].caps; + + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = get_rates(caps->rate_min, caps->rate_max); + stream->formats = caps->formats; + } + + if (dai->capture) { + stream = &dai_drv->capture; + caps = &dai->capconf[SND_SOC_TPLG_STREAM_CAPTURE].caps; + + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = get_rates(caps->rate_min, caps->rate_max); + stream->formats = caps->formats; + } + + dai_drv->dobj.index = tplg->index; + dai_drv->dobj.ops = tplg->ops; + dai_drv->dobj.type = SND_SOC_DOBJ_PCM; + list_add(&dai_drv->dobj.list, &tplg->comp->dobj_list); + + return snd_soc_add_dai(tplg->comp, dai_drv); +} + static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_pcm_dai *pcm_dai; struct snd_soc_dobj *dobj; int count = hdr->count; + int i; int ret; if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) @@ -1579,26 +1645,21 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, return -EINVAL; } - dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); - tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; - - dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL); - if (dobj == NULL) - return -ENOMEM; - - /* Call the platform driver call back to register the dais */ + /* pass control to driver for optional further init */ ret = soc_tplg_pcm_dai_load(tplg, pcm_dai, count); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: PCM DAI loading failed\n"); goto err; } - dobj->type = get_dobj_type(hdr, NULL); - dobj->pcm_dai.count = count; - dobj->pcm_dai.pd = pcm_dai; - dobj->ops = tplg->ops; - dobj->index = tplg->index; - list_add(&dobj->list, &tplg->comp->dobj_list); + /* register the dais */ + for (i = 0; i < count; i++) { + soc_tplg_pcm_dai_create(tplg, pcm_dai); + pcm_dai++; + } + + dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); + tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; return 0; err: -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 09/10] ASoC: Topology: change stream formats to bitwise flag
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The toplogy user space tool will generate this bitwise flag by using SNDRV_PCM_FORMAT_* exposed by asound.h, and the topology core will copy this flag when generating DAI streams. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69383b0..eb2ad59 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -191,7 +191,7 @@ struct snd_soc_tplg_ctl_hdr { struct snd_soc_tplg_stream_caps { __le32 size; /* in bytes of this structure */ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le64 formats; /* supported formats SNDRV_PCM_FORMAT_* */ __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ __le32 rate_min; /* min rate */ __le32 rate_max; /* max rate */ -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 07/10] ASoC: support adding a DAI dynamically
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> API snd_soc_add_dai() is defined to register a DAI dynamically, create its DAI widget, and notify the machine driver. This API can be used by the topology core to add DAIs dynamically. And a callback add_dai() is defined for the soc card. The machine driver can implement this callback. Adding a new DAI dynmaically will trigger this callback, and the machine driver can create relavant DAI link in the callback context. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 7e04ec0..2a474c8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1081,6 +1081,7 @@ struct snd_soc_card { int (*set_bias_level_post)(struct snd_soc_card *, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); + int (*add_dai)(struct snd_soc_card *, struct snd_soc_dai_driver*); long pmdown_time; @@ -1634,6 +1635,9 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, void snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +int snd_soc_add_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2818709..b330676 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2792,6 +2792,98 @@ err: return ret; } +static struct snd_soc_dai *soc_register_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv, + bool legacy_dai_naming) +{ + struct device *dev = component->dev; + struct snd_soc_dai *dai; + int ret; + + dev_dbg(dev, "ASoC: dynamically register DAI %s\n", dev_name(dev)); + + dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); + if (dai == NULL) { + ret = -ENOMEM; + goto err; + } + + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, dai_drv); + if (dai_drv->id) + dai->id = dai_drv->id; + else + dai->id = component->num_dai; + } + if (dai->name == NULL) { + kfree(dai); + ret = -ENOMEM; + goto err; + } + + dai->component = component; + dai->dev = dev; + dai->driver = dai_drv; + if (!dai->driver->ops) + dai->driver->ops = &null_dai_ops; + + list_add(&dai->list, &component->dai_list); + component->num_dai++; + + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); + return dai; + +err: + kfree(dai); + return NULL; +} + +/** + * snd_soc_add_dai - Add a DAI dynamically with the ASoC core + * + * @component: The component the DAIs are registered for + * @dai_drv: DAI driver to use for the DAI + */ +int snd_soc_add_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct snd_soc_card *card = component->card; + struct snd_soc_dai *dai; + int ret; + + lockdep_assert_held(&client_mutex); + dai = soc_register_dai(component, dai_drv, false); + if (!dai) + return -ENOMEM; + + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", ret); + } + + /* Notify the machine driver a new DAI is added, so the machine driver + * can add new DAI links in the callback context. + */ + if (card && card->add_dai) + card->add_dai(card, dai_drv); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai); + static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type type, int subseq) { -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 06/10] ASoC: bind new DAI links after probing components.
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Probing components can bring new DAI or DAI links based on the topology info. This patch finds the unbound DAI links and bind them. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f552fa0..2818709 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1169,6 +1169,19 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static bool soc_is_dai_link_bound(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link == dai_link) + return true; + } + + return false; +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -1763,6 +1776,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&client_mutex); @@ -1853,6 +1867,21 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* no longer need dummy runtimes */ soc_remove_dummy_pcm_runtimes(card); + /* Find new DAI links added during probing components and bind them. + * Components with topology may bring new DAIs and DAI links. + */ + list_for_each_entry(dai_link, &card->dai_link_list, list) { + if (dai_link->dummy || soc_is_dai_link_bound(card, dai_link)) + continue; + + ret = soc_init_dai_link(card, dai_link); + if (ret) + goto probe_dai_err; + ret = soc_bind_dai_link(card, dai_link); + if (ret) + goto probe_dai_err; + } + /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 05/10] ASoC: Add support for dummy DAI links and PCM runtimes
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Add dummy flag to a DAI link and PCM runtime. A dummy link will create a dummy PCM runtime. A machine driver can define dummy dai links to specify the expected platform and codec. And its dummy runtime is used to probe the platform and codec components, which can bring real DAI/DAI links based on topology. After components probing, all dummy PCM runtimes will be removed. Only non-dummy PCM runtimes will be assigned a valid number and create PCM streams. The ASoC core will find the platform for a dummy DAI link, assuming the cpu DAI can be dummy but the platform can bring the topology info. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 79891e2..7e04ec0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -970,6 +970,11 @@ struct snd_soc_dai_link { enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ + /* This DAI link is dummy. Machine drivers can define static dummy + * links to specify platform/codec with topology. + */ + bool dummy; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); @@ -1087,6 +1092,7 @@ struct snd_soc_card { struct list_head rtd_list; int num_rtd; + int num_rtd_dev; /* monotonic increase and exclude dummy ones */ /* optional codec specific configuration */ struct snd_soc_codec_conf *codec_conf; @@ -1181,6 +1187,7 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_state; #endif + bool dummy; unsigned int num; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 58eac57..f552fa0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -561,6 +561,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->card = card; rtd->dai_link = dai_link; + rtd->dummy = dai_link->dummy; rtd->codec_dais = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai *) * dai_link->num_codecs, @@ -583,10 +584,28 @@ static void soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { list_add_tail(&rtd->list, &card->rtd_list); - rtd->num = card->num_rtd; + + if (!rtd->dummy) { + rtd->num = card->num_rtd_dev; + card->num_rtd_dev++; + } + card->num_rtd++; } +static void soc_remove_dummy_pcm_runtimes(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd, *_rtd; + + list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + if (rtd->dummy) { + list_del(&rtd->list); + soc_free_pcm_runtime(card, rtd); + card->num_rtd--; + } + } +} + static void soc_remove_pcm_runtimes(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd, *_rtd; @@ -1007,6 +1026,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, } } + /* Dummy link can use dummy cpu DAI but set platform to get topology */ + if (dai_link->dummy && dai_link->platform_name) { + if (!soc_find_component(NULL, dai_link->platform_name)) { + dev_err(card->dev, "ASoC: Platform %s not registered\n", + dai_link->platform_name); + goto _err_defer; + } + } + /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; rtd->codec = rtd->codec_dai->codec; @@ -1250,6 +1278,10 @@ static int soc_init_dai_link(struct snd_soc_card *card, void snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* Dummy links should be static, so overlook them */ + if (dai_link->dummy) + return; + lockdep_assert_held(&client_mutex); list_add_tail(&dai_link->list, &card->dai_link_list); card->num_dai_links++; @@ -1818,6 +1850,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* no longer need dummy runtimes */ + soc_remove_dummy_pcm_runtimes(card); + /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { @@ -2516,6 +2551,7 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; + card->num_rtd_dev = 0; card->rtd_aux = devm_kzalloc(card->dev, sizeof(struct snd_soc_pcm_runtime) * -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 04/10] ASoC: implement DAI links in a list.
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Implement a dai link list for the soc card. And API snd_soc_add_dai_link() is defined for machine drivers to add DAI links dynamically based on topology. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index e2b7b0f..79891e2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1011,6 +1011,8 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + + struct list_head list; /* dai link list of the soc card */ }; struct snd_soc_codec_conf { @@ -1078,8 +1080,11 @@ struct snd_soc_card { long pmdown_time; /* CPU <--> Codec DAI links */ - struct snd_soc_dai_link *dai_link; - int num_links; + struct snd_soc_dai_link *dai_link; /* predefined links only */ + int num_links; /* predefined links only */ + struct list_head dai_link_list; /* all links except the dummy ones */ + int num_dai_links; + struct list_head rtd_list; int num_rtd; @@ -1619,6 +1624,9 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link); +void snd_soc_add_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ddfdd1d..58eac57 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1235,6 +1235,27 @@ static int soc_init_dai_link(struct snd_soc_card *card, return 0; } +/** + * snd_soc_add_dai_link - Add a DAI link dynamically + * @card: The ASoC card to which the DAI link is added + * @dai_link: The new DAI link to add + * + * This function adds a DAI link to the ASoC card, and creates a ASoC + * runtime for the link. + * + * Note: For adding DAI links dynamically by machine drivers based on + * the topology info when probing the platform component. And machine + * drivers can still define static DAI links in dai_link array. + */ +void snd_soc_add_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + lockdep_assert_held(&client_mutex); + list_add_tail(&dai_link->list, &card->dai_link_list); + card->num_dai_links++; +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai_link); + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -1729,6 +1750,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } + /* all predefined resources are ready, store non-dummy links */ + for (i = 0; i < card->num_links; i++) + snd_soc_add_dai_link(card, card->dai_link+i); + /* initialize the register cache for each available codec */ list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) @@ -2486,6 +2511,9 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); + INIT_LIST_HEAD(&card->dai_link_list); + card->num_dai_links = 0; + INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 03/10] ASoC: soc_bind_dai_link() change 2nd argument to DAI link pointer
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Just code refactoring, to reuse it if new DAI Links are added later based on topology in component probing phase. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e9e8956..ddfdd1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -961,9 +961,9 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } -static int soc_bind_dai_link(struct snd_soc_card *card, int num) +static int soc_bind_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; @@ -972,7 +972,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) const char *platform_name; int i; - dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (snd_soc_get_pcm_runtime(card, dai_link->name)) { dev_dbg(card->dev, "ASoC: dai link %s already bound\n", @@ -1717,7 +1717,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* bind DAIs */ for (i = 0; i < card->num_links; i++) { - ret = soc_bind_dai_link(card, i); + ret = soc_bind_dai_link(card, &card->dai_link[i]); if (ret != 0) goto base_error; } -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 01/10] ASoC: change the PCM runtime array to a list
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Currently the number of DAI links is statically defined by the machine driver at build time using an array. This makes it difficult to shrink/ grow the number of DAI links at runtime in order to reflect any changes in topology. We can change the DAI link array in the core to a list so that PCMs and FE DAI links can be added and deleted at runtime to reflect changes in use case and DSP topology. The machine driver can still register DAI links as an array. As the 1st step, this patch change the PCM runtime array to a list. A new PCM runtime is added to the list when a DAI link is bound successfully. Later patches will further implement the DAI link list. More: - define snd_soc_new/free_pcm_runtime() to create/free a runtime. - define soc_add_pcm_runtime() to add a runtime to the rtd list. - define soc_remove_pcm_runtimes() to clean up the runtime list. - traverse the rtd list to probe the link components and dais. - Add a field "num" to PCM runtime struct, used to specify the device number when creating the pcm device, and for a simple card to access its dai_props array. - The following 3rd party machine/platform drivers either iterate the rtd list or use snd_soc_get_pcm_runtime() to access a rtd: intel/atom/sst-mfld-platform-pcm.c intel/boards/cht_bsw_rt5672 intel/boards/cht_bsw_rt5645 fsl/imx-wm8962.c fsl/fsl-asoc-card.c samsung/bells.c samsung/littlemill.c samsung/odroidx2_max98090.c samsung/snow.c samsung/speyside.c samsung/tobermory.c pxa/mioa701_wm9713.c tegra/tegra_wm8903.c Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 4cef20e..e2b7b0f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1080,7 +1080,7 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; - struct snd_soc_pcm_runtime *rtd; + struct list_head rtd_list; int num_rtd; /* optional codec specific configuration */ @@ -1175,6 +1175,9 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_root; struct dentry *debugfs_dpcm_state; #endif + + unsigned int num; /* 0-based and monotonic increasing */ + struct list_head list; /* rtd list of the soc card */ }; /* mixer control */ diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..b80cf09 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -211,12 +211,15 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; unsigned int pll_out; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; @@ -383,11 +386,14 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, static int fsl_asoc_card_late_probe(struct snd_soc_card *card) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index b38b98c..201a70d 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -69,13 +69,16 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct imx_priv *priv = &card_priv; struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; @@ -135,12 +138,15 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, static int imx_wm8962_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct imx_priv *priv = &card_priv; struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ff76d4..95a2f91 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -45,7 +45,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + &priv->dai_props[rtd->num]; int ret; ret = clk_prepare_enable(dai_props->cpu_dai.clk); @@ -64,7 +64,7 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + &priv->dai_props[rtd->num]; clk_disable_unprepare(dai_props->cpu_dai.clk); @@ -78,8 +78,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + struct simple_dai_props *dai_props = &priv->dai_props[rtd->num]; unsigned int mclk, mclk_fs = 0; int ret = 0; @@ -172,10 +171,9 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; struct simple_dai_props *dai_props; - int num, ret; + int ret; - num = rtd - rtd->card->rtd; - dai_props = &priv->dai_props[num]; + dai_props = &priv->dai_props[rtd->num]; ret = __asoc_simple_card_dai_init(codec, &dai_props->codec_dai); if (ret < 0) return ret; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e501..5ae8ae2 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -777,15 +777,15 @@ static int sst_platform_remove(struct platform_device *pdev) static int sst_soc_prepare(struct device *dev) { struct sst_data *drv = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; /* suspend all pcms first */ snd_soc_suspend(drv->soc_card->dev); snd_soc_poweroff(drv->soc_card->dev); /* set the SSPs to idle */ - for (i = 0; i < drv->soc_card->num_rtd; i++) { - struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { send_ssp_cmd(dai, dai->name, 0); @@ -799,11 +799,11 @@ static int sst_soc_prepare(struct device *dev) static void sst_soc_complete(struct device *dev) { struct sst_data *drv = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; /* restart SSPs */ - for (i = 0; i < drv->soc_card->num_rtd; i++) { - struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { sst_handle_vb_timer(dai, true); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..fade1a2 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -47,12 +47,9 @@ struct cht_mc_private { static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { - int i; - - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; + struct snd_soc_pcm_runtime *rtd; - rtd = card->rtd + i; + list_for_each_entry(rtd, &card->rtd_list, list) { if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, strlen(CHT_CODEC_DAI))) return rtd->codec_dai; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..fee43bd 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -46,12 +46,9 @@ static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; - - rtd = card->rtd + i; + list_for_each_entry(rtd, &card->rtd_list, list) { if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, strlen(CHT_CODEC_DAI))) return rtd->codec_dai; diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index a9615a57..4f6cafe 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -81,8 +81,12 @@ static int rear_amp_power(struct snd_soc_codec *codec, int power) static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kctl, int event) { - struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec; + struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); } diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index e5f05e6..3dd246f 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -58,11 +58,16 @@ static int bells_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[DAI_DSP_CODEC].codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_codec *codec; struct bells_drvdata *bells = card->drvdata; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec_dai = rtd->codec_dai; + codec = codec_dai->codec; + if (dapm->dev != codec_dai->dev) return 0; @@ -99,11 +104,16 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[DAI_DSP_CODEC].codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_codec *codec; struct bells_drvdata *bells = card->drvdata; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec_dai = rtd->codec_dai; + codec = codec_dai->codec; + if (dapm->dev != codec_dai->dev) return 0; @@ -137,14 +147,22 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, static int bells_late_probe(struct snd_soc_card *card) { struct bells_drvdata *bells = card->drvdata; - struct snd_soc_codec *wm0010 = card->rtd[DAI_AP_DSP].codec; - struct snd_soc_codec *codec = card->rtd[DAI_DSP_CODEC].codec; - struct snd_soc_dai *aif1_dai = card->rtd[DAI_DSP_CODEC].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *wm0010; + struct snd_soc_codec *codec; + struct snd_soc_dai *aif1_dai; struct snd_soc_dai *aif2_dai; struct snd_soc_dai *aif3_dai; struct snd_soc_dai *wm9081_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_AP_DSP].name); + wm0010 = rtd->codec; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec = rtd->codec; + aif1_dai = rtd->codec_dai; + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, ARIZONA_CLK_SRC_FLL1, bells->sysclk_rate, @@ -181,7 +199,8 @@ static int bells_late_probe(struct snd_soc_card *card) return ret; } - aif2_dai = card->rtd[DAI_CODEC_CP].cpu_dai; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_CODEC_CP].name); + aif2_dai = rtd->cpu_dai; ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret != 0) { @@ -192,8 +211,9 @@ static int bells_late_probe(struct snd_soc_card *card) if (card->num_rtd == DAI_CODEC_SUB) return 0; - aif3_dai = card->rtd[DAI_CODEC_SUB].cpu_dai; - wm9081_dai = card->rtd[DAI_CODEC_SUB].codec_dai; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_CODEC_SUB].name); + aif3_dai = rtd->cpu_dai; + wm9081_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret != 0) { diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 31a820e..7cb204e 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,9 +23,13 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif1_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + aif1_dai = rtd->codec_dai; + if (dapm->dev != aif1_dai->dev) return 0; @@ -66,9 +70,13 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif1_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + aif1_dai = rtd->codec_dai; + if (dapm->dev != aif1_dai->dev) return 0; @@ -168,9 +176,13 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_card *card = w->dapm->card; - struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif2_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + aif2_dai = rtd->cpu_dai; + switch (event) { case SND_SOC_DAPM_PRE_PMU: ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, @@ -245,11 +257,19 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + struct snd_soc_dai *aif1_dai; + struct snd_soc_dai *aif2_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; + aif1_dai = rtd->codec_dai; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + aif2_dai = rtd->cpu_dai; + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 596f118..0421727 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -25,10 +25,15 @@ static struct snd_soc_dai_link odroidx2_dai[]; static int odroidx2_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + cpu_dai = rtd->cpu_dai; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..0969577 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -35,10 +35,15 @@ static struct snd_soc_dai_link snow_dai[] = { static int snow_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + cpu_dai = rtd->cpu_dai; + /* Set the MCLK rate for the codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, FIN_PLL_RATE, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index d1ae21c..083ef5e 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -25,9 +25,13 @@ static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -57,9 +61,13 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 85ccfb7..3310eda 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -23,9 +23,13 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -62,9 +66,13 @@ static int tobermory_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -170,10 +178,15 @@ static struct snd_soc_jack_pin tobermory_headset_pins[] = { static int tobermory_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; + codec_dai = rtd->codec_dai; + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b63a03..fb78674 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -537,26 +537,76 @@ static inline void snd_soc_debugfs_exit(void) struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_links; i++) { - if (card->rtd[i].dai_link->no_pcm && - !strcmp(card->rtd[i].dai_link->name, dai_link)) - return card->rtd[i].pcm->streams[stream].substream; + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link->no_pcm && + !strcmp(rtd->dai_link->name, dai_link)) + return rtd->pcm->streams[stream].substream; } dev_dbg(card->dev, "ASoC: failed to find dai link %s\n", dai_link); return NULL; } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( + struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_pcm_runtime *rtd; + + rtd = devm_kzalloc(card->dev, + sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); + if (!rtd) + return NULL; + + rtd->card = card; + rtd->dai_link = dai_link; + rtd->codec_dais = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai *) * + dai_link->num_codecs, + GFP_KERNEL); + if (!rtd->codec_dais) + return NULL; + + return rtd; +} + +static void soc_free_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + if (rtd->codec_dai) + devm_kfree(card->dev, rtd->codec_dais); + devm_kfree(card->dev, rtd); +} + +static void soc_add_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + list_add_tail(&rtd->list, &card->rtd_list); + rtd->num = card->num_rtd; + card->num_rtd++; +} + +static void soc_remove_pcm_runtimes(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd, *_rtd; + + list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + list_del(&rtd->list); + soc_free_pcm_runtime(card, rtd); + } + + card->num_rtd = 0; +} + struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_links; i++) { - if (!strcmp(card->rtd[i].dai_link->name, dai_link)) - return &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { + if (!strcmp(rtd->dai_link->name, dai_link)) + return rtd; } dev_dbg(card->dev, "ASoC: failed to find rtd %s\n", dai_link); return NULL; @@ -578,7 +628,8 @@ int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; - int i, j; + struct snd_soc_pcm_runtime *rtd; + int i; /* If the card is not initialized yet there is nothing to do */ if (!card->instantiated) @@ -595,13 +646,13 @@ int snd_soc_suspend(struct device *dev) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DACs */ - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - for (j = 0; j < card->rtd[i].num_codecs; j++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -610,20 +661,20 @@ int snd_soc_suspend(struct device *dev) } /* suspend all pcms */ - for (i = 0; i < card->num_rtd; i++) { - if (card->rtd[i].dai_link->ignore_suspend) + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link->ignore_suspend) continue; - snd_pcm_suspend_all(card->rtd[i].pcm); + snd_pcm_suspend_all(rtd->pcm); } if (card->suspend_pre) card->suspend_pre(card); - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control) @@ -631,19 +682,19 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - for (i = 0; i < card->num_rtd; i++) - flush_delayed_work(&card->rtd[i].delayed_work); + list_for_each_entry(rtd, &card->rtd_list, list) + flush_delayed_work(&rtd->delayed_work); - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_SUSPEND); - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -690,10 +741,10 @@ int snd_soc_suspend(struct device *dev) } } - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control) @@ -717,8 +768,9 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); + struct snd_soc_pcm_runtime *rtd; struct snd_soc_codec *codec; - int i, j; + int i; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -733,10 +785,10 @@ static void soc_resume_deferred(struct work_struct *work) card->resume_pre(card); /* resume control bus DAIs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && cpu_dai->driver->bus_control) @@ -751,28 +803,28 @@ static void soc_resume_deferred(struct work_struct *work) } } - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - for (j = 0; j < card->rtd[i].num_codecs; j++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -780,10 +832,10 @@ static void soc_resume_deferred(struct work_struct *work) } } - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control) @@ -808,15 +860,14 @@ int snd_soc_resume(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); bool bus_control = false; - int i; + struct snd_soc_pcm_runtime *rtd; /* If the card is not initialized yet there is nothing to do */ if (!card->instantiated) return 0; /* activate pins from sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; @@ -837,8 +888,8 @@ int snd_soc_resume(struct device *dev) * have that problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; bus_control |= cpu_dai->driver->bus_control; } if (bus_control) { @@ -913,16 +964,26 @@ static struct snd_soc_dai *snd_soc_find_dai( static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; - struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai **codec_dais; struct snd_soc_platform *platform; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); + if (snd_soc_get_pcm_runtime(card, dai_link->name)) { + dev_dbg(card->dev, "ASoC: dai link %s already bound\n", + dai_link->name); + return 0; + } + + rtd = soc_new_pcm_runtime(card, dai_link); + if (!rtd) + return -ENOMEM; + cpu_dai_component.name = dai_link->cpu_name; cpu_dai_component.of_node = dai_link->cpu_of_node; cpu_dai_component.dai_name = dai_link->cpu_dai_name; @@ -930,18 +991,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); - return -EPROBE_DEFER; + goto _err_defer; } rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + codec_dais = rtd->codec_dais; for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); - return -EPROBE_DEFER; + goto _err_defer; } } @@ -973,9 +1035,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - card->num_rtd++; - + soc_add_pcm_runtime(card, rtd); return 0; + +_err_defer: + soc_free_pcm_runtime(card, rtd); + return -EPROBE_DEFER; } static void soc_remove_component(struct snd_soc_component *component) @@ -1014,9 +1079,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) } } -static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) +static void soc_remove_link_dais(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; int i; /* unregister the rtd device */ @@ -1032,10 +1097,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) soc_remove_dai(rtd->cpu_dai, order); } -static void soc_remove_link_components(struct snd_soc_card *card, int num, - int order) +static void soc_remove_link_components(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_component *component; @@ -1061,21 +1125,20 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, static void soc_remove_dai_links(struct snd_soc_card *card) { - int dai, order; + int order; + struct snd_soc_pcm_runtime *rtd; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_link_dais(card, dai, order); + list_for_each_entry(rtd, &card->rtd_list, list) + soc_remove_link_dais(card, rtd, order); } for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_link_components(card, dai, order); + list_for_each_entry(rtd, &card->rtd_list, list) + soc_remove_link_components(card, rtd, order); } - - card->num_rtd = 0; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1220,10 +1283,10 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return 0; } -static int soc_probe_link_components(struct snd_soc_card *card, int num, +static int soc_probe_link_components(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_component *component; int i, ret; @@ -1319,15 +1382,15 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, return 0; } -static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) +static int soc_probe_link_dais(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", - card->name, num, order); + card->name, rtd->num, order); /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1372,7 +1435,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = soc_new_compress(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1382,7 +1445,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, num); + ret = soc_new_pcm(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1552,6 +1615,7 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; + struct snd_soc_pcm_runtime *rtd; int ret, i, order; mutex_lock(&client_mutex); @@ -1624,8 +1688,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_link_components(card, i, order); + list_for_each_entry(rtd, &card->rtd_list, list) { + ret = soc_probe_link_components(card, rtd, order); if (ret < 0) { dev_err(card->dev, "ASoC: failed to instantiate card %d\n", @@ -1638,8 +1702,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_link_dais(card, i, order); + list_for_each_entry(rtd, &card->rtd_list, list) { + ret = soc_probe_link_dais(card, rtd, order); if (ret < 0) { dev_err(card->dev, "ASoC: failed to instantiate card %d\n", @@ -1733,6 +1797,7 @@ card_probe_error: snd_card_free(card->snd_card); base_error: + soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -1763,13 +1828,12 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { + struct snd_soc_pcm_runtime *rtd; int i; /* make sure any delayed work runs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - } /* remove auxiliary devices */ for (i = 0; i < card->num_aux_devs; i++) @@ -1777,6 +1841,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); soc_cleanup_card_debugfs(card); @@ -1803,29 +1868,26 @@ static int soc_remove(struct platform_device *pdev) int snd_soc_poweroff(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; if (!card->instantiated) return 0; /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - } snd_soc_dapm_shutdown(card); /* deactivate pins to sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; + int i; pinctrl_pm_select_sleep_state(cpu_dai->dev); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; pinctrl_pm_select_sleep_state(codec_dai->dev); } } @@ -2337,6 +2399,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, int snd_soc_register_card(struct snd_soc_card *card) { int i, j, ret; + struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2408,25 +2471,15 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); - card->rtd = devm_kzalloc(card->dev, + INIT_LIST_HEAD(&card->rtd_list); + card->num_rtd = 0; + + card->rtd_aux = devm_kzalloc(card->dev, sizeof(struct snd_soc_pcm_runtime) * - (card->num_links + card->num_aux_devs), + card->num_aux_devs, GFP_KERNEL); - if (card->rtd == NULL) + if (card->rtd_aux == NULL) return -ENOMEM; - card->num_rtd = 0; - card->rtd_aux = &card->rtd[card->num_links]; - - for (i = 0; i < card->num_links; i++) { - card->rtd[i].card = card; - card->rtd[i].dai_link = &card->dai_link[i]; - card->rtd[i].codec_dais = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai *) * - (card->rtd[i].dai_link->num_codecs), - GFP_KERNEL); - if (card->rtd[i].codec_dais == NULL) - return -ENOMEM; - } for (i = 0; i < card->num_aux_devs; i++) card->rtd_aux[i].card = card; @@ -2442,8 +2495,7 @@ int snd_soc_register_card(struct snd_soc_card *card) return ret; /* deactivate pins to sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..6d1f066 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3861,13 +3861,10 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = card->rtd; - int i; + struct snd_soc_pcm_runtime *rtd; /* for each BE DAI link... */ - for (i = 0; i < card->num_rtd; i++) { - rtd = &card->rtd[i]; - + list_for_each_entry(rtd, &card->rtd_list, list) { /* * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e4b9d..0246019 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1166,11 +1166,10 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; - int i, j; + int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < card->num_links; i++) { - be = &card->rtd[i]; + list_for_each_entry(be, &card->rtd_list, list) { if (!be->dai_link->no_pcm) continue; @@ -1178,16 +1177,15 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->playback_widget == widget) return be; - for (j = 0; j < be->num_codecs; j++) { - struct snd_soc_dai *dai = be->codec_dais[j]; + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; if (dai->playback_widget == widget) return be; } } } else { - for (i = 0; i < card->num_links; i++) { - be = &card->rtd[i]; + list_for_each_entry(be, &card->rtd_list, list) { if (!be->dai_link->no_pcm) continue; @@ -1195,8 +1193,8 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->capture_widget == widget) return be; - for (j = 0; j < be->num_codecs; j++) { - struct snd_soc_dai *dai = be->codec_dais[j]; + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; if (dai->capture_widget == widget) return be; } @@ -2296,12 +2294,12 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) */ int soc_dpcm_runtime_update(struct snd_soc_card *card) { - int i, old, new, paths; + struct snd_soc_pcm_runtime *fe; + int old, new, paths; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(fe, &card->rtd_list, list) { struct snd_soc_dapm_widget_list *list; - struct snd_soc_pcm_runtime *fe = &card->rtd[i]; /* make sure link is FE */ if (!fe->dai_link->dynamic) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2160400..e485278 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -199,7 +199,8 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) static int tegra_wm8903_remove(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_pcm_runtime *rtd = + snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); -- 1.9.1

1 0

Re: [alsa-devel] [SOLVED] [pulseaudio-discuss] No sound Lubuntu 14.02
by Raymond Yau 08 Aug '15

08 Aug '15

>>>> >> > >>>> >> > sl@max:~$ lspci -nnk | grep -iA2 audio >>>> >> > 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller [8086:2668] (rev 04) >>>> >> > Subsystem: QUANTA Computer Inc Device [152d:0748] >>>> >> > Kernel driver in use: snd_hda_intel >>>> >> > >>>> >> > sl@max:~$ aplay -l >>>> >> > **** Liste der Hardware-Geräte (PLAYBACK) **** >>>> >> > Karte 0: Intel [HDA Intel], Gerät 6: Si3054 Modem [Si3054 Modem] >>>> >> > Sub-Geräte: 1/1 >>>> >> > Sub-Gerät #0: subdevice #0 >>>> >> > >>>> >> >>>> >> https://lists.fedoraproject.org/pipermail/test/2014-November/123662.html >>>> >> >>>> >> If bios did not setup pin default of alc880 codec, you need to find out the pin of hp and mic jacks by hdajacksensetest , speaker and internal mic need to find out by trial and err with hdajackretask >>>> >> >>>> > >>>> > This is the output of hda-jack-sense-test. What do you recommend? >>>> > >>>> > root@max:~# hda-jack-sense-test >>>> > Pin 0x14 (Unknown Line Out): present = No >>>> > Pin 0x15 (Unknown Line Out): present = No >>>> > Pin 0x16 (Unknown Line Out): present = No >>>> > Pin 0x17 (Unknown Line Out): present = No >>>> > Pin 0x18 (Unknown Line Out): present = No >>>> > Pin 0x19 (Unknown Line Out): present = No >>>> > Pin 0x1a (Unknown Line Out): present = No >>>> > Pin 0x1b (Unknown Line Out): present = No >>>> > Pin 0x1c (Unknown Line Out): present = No >>>> > Pin 0x1d (Unknown Line Out): present = No >>>> > Pin 0x1e (Unknown Line Out): present = No >>>> > Pin 0x1f (Unknown Line Out): present = No >>>> > >>>> >>>> Post the output of alsa-info.sh >>>> >>>> You have to use >>>> >>>> hdajacksensetest -a >>> >>> It gives the same output. >>>> >>>> Did you plug and unplug headphone or mic jack during test ? >>> >>> No. >> >> you have to post output of >> >> hdajacksensetest -a >> >> when you plug headphone , unplug headphone , plug mic and unplug mic >> >> >> >>>> >>>> Since pin default of all pin complex are zero which is Line Out >>> >>> It is a laptop. I want to use its internal speakers, if that info helps. >> >> All pins have to be fixed by hdajackretask >> >> !!Sysfs Files >> !!----------- >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x00000000 >> 0x15 0x00000000 >> 0x16 0x00000000 >> 0x17 0x00000000 >> 0x18 0x00000000 >> 0x19 0x00000000 >> 0x1a 0x00000000 >> 0x1b 0x00000000 >> 0x1c 0x00000000 >> 0x1d 0x00000000 >> 0x1e 0x01000000 >> 0x1f 0x00000000 >> >> >>>> >>>> Any redundant pins must set to [N/A] >>>> >>>> alc880 only have models for desktop >>>> >>>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >>>> >>>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/… >>> >>> >> >> >> In the past, there was a "test" model which setup all pins >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> you can try >> >> options snd-hda-intel model=6stack-dig >> >> to find out whether the speaker is node 0x14 and Headphone and 0x1b >> you have to manually find front mic, rear mic and line in is your laptop internal mic and mic jacks >> >> >> Do windows driver contain info about the pins since it also need to setup the pin complex too ? >> >> >> >> sl@max:~$ cat /tmp/alsa-info.txt.5lh8RkGutE >> upload=true&script=true&cardinfo= >> !!################################ >> !!ALSA Information Script v 0.4.64 >> !!################################ >> >> !!Script ran on: Fri Aug 7 11:57:07 UTC 2015 >> >> >> !!Linux Distribution >> !!------------------ >> >> Ubuntu 14.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL=" http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" >> >> >> !!DMI Information >> !!--------------- >> >> Manufacturer: MAXDATA >> Product Name: PRO8100IS58 >> Product Version: 5123660003 >> Firmware Version: M3B61 >> >> >> !!Kernel Information >> !!------------------ >> >> Kernel release: 3.16.0-45-generic >> Operating System: GNU/Linux >> Architecture: i686 >> Processor: i686 >> SMP Enabled: Yes >> >> >> !!ALSA Version >> !!------------ >> >> Driver version: k3.16.0-45-generic >> Library version: 1.0.27.2 >> Utilities version: 1.0.27.2 >> >> >> !!Loaded ALSA modules >> !!------------------- >> >> snd_hda_intel >> >> >> !!Sound Servers on this system >> !!---------------------------- >> >> Pulseaudio: >> Installed - Yes (/usr/bin/pulseaudio) >> Running - Yes >> >> >> !!Soundcards recognised by ALSA >> !!----------------------------- >> >> 0 [Intel ]: HDA-Intel - HDA Intel >> HDA Intel at 0xc0000000 irq 46 >> >> >> !!PCI Soundcards installed in the system >> !!-------------------------------------- >> >> 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) >> >> >> !!Advanced information - PCI Vendor/Device/Subsystem ID's >> !!------------------------------------------------------- >> >> 00:1b.0 0403: 8086:2668 (rev 04) >> Subsystem: 152d:0748 >> -- >> Prefetchable memory behind bridge: 0000000040000000-00000000401fffff >> Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- >> >> >> >> >> !!HDA-Intel Codec information >> !!--------------------------- >> --startcollapse-- >> >> Codec: Realtek ALC880 >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x10ec0880 >> Subsystem Id: 0x08800000 >> Revision Id: 0x100800 >> No Modem Function Group found >> Default PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Default Amp-In caps: N/A >> Default Amp-Out caps: N/A >> State of AFG node 0x01: >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 >> IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 >> IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 >> Node 0x02 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x03 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x04 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x05 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital >> Converter: stream=0, channel=0 >> Digital: >> Digital category: 0x0 >> IEC Coding Type: 0x0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x1e]: 16 20 24 32 >> formats [0x1]: PCM >> Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 7 >> 0x18* 0x19 0x1a 0x1b 0x1c 0x14 0x15 >> Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 7 >> 0x18 0x19 0x1a* 0x1b 0x1c 0x14 0x15 >> Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 10 >> 0x18 0x19 0x1a 0x1b 0x1c* 0x0b 0x14 0x15 0x16 0x17 >> Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> Digital: >> Digital category: 0x0 >> IEC Coding Type: 0x0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x1e]: 16 20 24 32 >> formats [0x1]: PCM >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x1f >> Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >> Control: name="Beep Playback Volume", index=0, device=0 >> ControlAmp: chs=3, dir=In, idx=5, ofs=0 >> Control: name="Beep Playback Switch", index=0, device=0 >> ControlAmp: chs=3, dir=In, idx=5, ofs=0 >> Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 >> Amp-In vals: [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0x80 0x80] [0xa3 0xa3] [0xa3 0xa3] >> Connection: 8 >> 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 >> Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x02 0x0b >> Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x03 0x0b >> Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x04 0x0b >> Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x05 0x0b >> Node 0x10 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x11 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x12 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x13 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0c >> Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0d >> Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0e >> Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0f >> Node 0x18 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x21: IN VREF_50 >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x10 >> Node 0x19 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x11 >> Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x12 >> Node 0x1b [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x13 >> Node 0x1c [Pin Complex] wcaps 0x400001: Stereo >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x1d [Pin Complex] wcaps 0x400000: Mono >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital >> Control: name="Line Out Phantom Jack", index=0, device=0 >> Pincap 0x00000010: OUT >> Pin Default 0x01000000: [Jack] Line Out at Ext Rear >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Connection: 1 >> 0x06 >> Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono >> Processing caps: benign=0, ncoeff=10 >> Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono >> Volume-Knob: delta=0, steps=64, direct=0, val=25 >> Unsolicited: tag=00, enabled=0 >> Connection: 0 >> Codec: Motorola Si3054 >> Address: 1 >> MFG Function Id: 0x2 (unsol 1) >> Vendor Id: 0x10573055 >> Subsystem Id: 0x10573055 >> Revision Id: 0x100700 >> Modem Function Group: 0x1 >> --endcollapse-- >> >> >> !!ALSA Device nodes >> !!----------------- >> >> crw-rw----+ 1 root audio 116, 2 Aug 7 12:03 /dev/snd/controlC0 >> crw-rw----+ 1 root audio 116, 5 Aug 7 12:03 /dev/snd/hwC0D0 >> crw-rw----+ 1 root audio 116, 6 Aug 7 12:03 /dev/snd/hwC0D1 >> crw-rw----+ 1 root audio 116, 4 Aug 7 12:03 /dev/snd/pcmC0D6c >> crw-rw----+ 1 root audio 116, 3 Aug 7 12:03 /dev/snd/pcmC0D6p >> crw-rw----+ 1 root audio 116, 1 Aug 7 12:03 /dev/snd/seq >> crw-rw----+ 1 root audio 116, 33 Aug 7 12:03 /dev/snd/timer >> >> /dev/snd/by-path: >> total 0 >> drwxr-xr-x 2 root root 60 Aug 7 12:03 . >> drwxr-xr-x 3 root root 200 Aug 7 12:03 .. >> lrwxrwxrwx 1 root root 12 Aug 7 12:03 pci-0000:00:1b.0 -> ../controlC0 >> >> >> !!Aplay/Arecord output >> !!-------------------- >> >> APLAY >> >> **** List of PLAYBACK Hardware Devices **** >> card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> !!Amixer output >> !!------------- >> >> !!-------Mixer controls for card 0 [Intel] >> >> Card hw:0 'Intel'/'HDA Intel at 0xc0000000 irq 46' >> Mixer name : 'Realtek ALC880' >> Components : 'HDA:10ec0880,08800000,00100800 HDA:10573055,10573055,00100700' >> Controls : 8 >> Simple ctrls : 4 >> Simple mixer control 'PCM',0 >> Capabilities: pvolume >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 255 >> Mono: >> Front Left: Playback 255 [100%] [0.00dB] >> Front Right: Playback 255 [100%] [0.00dB] >> Simple mixer control 'Beep',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 65 >> Mono: >> Front Left: Playback 0 [0%] [-35.00dB] [off] >> Front Right: Playback 0 [0%] [-35.00dB] [off] >> Simple mixer control 'Caller ID',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> Simple mixer control 'Off-hook',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> >> >> !!Alsactl output >> !!-------------- >> >> --startcollapse-- >> state.Intel { >> control.1 { >> iface CARD >> name 'Line Out Phantom Jack' >> value true >> comment { >> access read >> type BOOLEAN >> count 1 >> } >> } >> control.2 { >> iface MIXER >> name 'Beep Playback Volume' >> value.0 0 >> value.1 0 >> comment { >> access 'read write' >> type INTEGER >> count 2 >> range '0 - 65' >> dbmin -3500 >> dbmax 3000 >> dbvalue.0 -3500 >> dbvalue.1 -3500 >> } >> } >> control.3 { >> iface MIXER >> name 'Beep Playback Switch' >> value.0 false >> value.1 false >> comment { >> access 'read write' >> type BOOLEAN >> count 2 >> } >> } >> control.4 { >> iface MIXER >> name 'Off-hook Switch' >> value false >> comment { >> access 'read write' >> type BOOLEAN >> count 1 >> } >> } >> control.5 { >> iface MIXER >> name 'Caller ID Switch' >> value false >> comment { >> access 'read write' >> type BOOLEAN >> count 1 >> } >> } >> control.6 { >> iface PCM >> device 6 >> name 'Playback Channel Map' >> value 0 >> comment { >> access read >> type INTEGER >> count 1 >> range '0 - 36' >> } >> } >> control.7 { >> iface PCM >> device 6 >> name 'Capture Channel Map' >> value 0 >> comment { >> access read >> type INTEGER >> count 1 >> range '0 - 36' >> } >> } >> control.8 { >> iface MIXER >> name 'PCM Playback Volume' >> value.0 255 >> value.1 255 >> comment { >> access 'read write user' >> type INTEGER >> count 2 >> range '0 - 255' >> tlv '0000000100000008ffffec1400000014' >> dbmin -5100 >> dbmax 0 >> dbvalue.0 0 >> dbvalue.1 0 >> } >> } >> } >> --endcollapse-- >> >> >> >> >> !!Sysfs Files >> !!----------- >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x00000000 >> 0x15 0x00000000 >> 0x16 0x00000000 >> 0x17 0x00000000 >> 0x18 0x00000000 >> 0x19 0x00000000 >> 0x1a 0x00000000 >> 0x1b 0x00000000 >> 0x1c 0x00000000 >> 0x1d 0x00000000 >> 0x1e 0x01000000 >> 0x1f 0x00000000 >> >> /sys/class/sound/hwC0D0/driver_pin_configs: >> >> /sys/class/sound/hwC0D0/user_pin_configs: >> >> /sys/class/sound/hwC0D0/init_verbs: >> >> /sys/class/sound/hwC0D0/hints: >> >> /sys/class/sound/hwC0D1/init_pin_configs: >> >> /sys/class/sound/hwC0D1/driver_pin_configs: >> >> /sys/class/sound/hwC0D1/user_pin_configs: >> >> /sys/class/sound/hwC0D1/init_verbs: >> >> /sys/class/sound/hwC0D1/hints: >> >> >> !!ALSA/HDA dmesg >> !!-------------- >> >> [ 29.335966] ipw2200: Detected geography ZZE (13 802.11bg channels, 19 802.11a channels) >> [ 29.378618] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X >> [ 29.673379] sound hdaudioC0D0: autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line >> [ 29.673387] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) >> [ 29.673390] sound hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) >> [ 29.673393] sound hdaudioC0D0: mono: mono_out=0x1e >> [ 29.673396] sound hdaudioC0D0: inputs: >> [ 30.010420] Bluetooth: RFCOMM TTY layer initialized >> >> >> > > > After figuring out what lines are used for which device, I was able > to set config. Now it is working. > Can you post out the correct pin config for your laptop ? Do auto mute and auto mic selection work as expected ? Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital >> Control: name="Line Out Phantom Jack", index=0, device=0 >> Pincap 0x00000010: OUT >> Pin Default 0x01000000: [Jack] Line Out at Ext Rear >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Connection: 1 >> 0x06 There is a bug in hda_auto_parser which treat invalid digital out pin as mono out(line out) and create Line Out Phantom Jack

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[alsa-devel] [PATCH] topology: treat all DAPM controls types the same when copying
by Liam Girdwood 07 Aug '15

07 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Copy all DAPM controls types using the same method. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/dapm.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 7e26ea0..a0a8b86 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -107,8 +107,6 @@ static int tplg_parse_dapm_enums(snd_config_t *cfg, struct tplg_elem *elem) static int copy_dapm_control(struct tplg_elem *elem, struct tplg_elem *ref) { struct snd_soc_tplg_dapm_widget *widget = elem->widget; - struct snd_soc_tplg_mixer_control *mixer_ctrl = ref->mixer_ctrl; - struct snd_soc_tplg_enum_control *enum_ctrl = ref->enum_ctrl; tplg_dbg("Control '%s' used by '%s'\n", ref->id, elem->id); tplg_dbg("\tparent size: %d + %d -> %d, priv size -> %d\n", @@ -121,13 +119,10 @@ static int copy_dapm_control(struct tplg_elem *elem, struct tplg_elem *ref) elem->widget = widget; - /* copy new widget at the end */ - if (ref->type == OBJECT_TYPE_MIXER) - memcpy((void*)widget + elem->size, mixer_ctrl, ref->size); - else if (ref->type == OBJECT_TYPE_ENUM) - memcpy((void*)widget + elem->size, enum_ctrl, ref->size); - + /* append the control to the end of the widget */ + memcpy((void*)widget + elem->size, ref->obj, ref->size); elem->size += ref->size; + widget->num_kcontrols++; ref->compound_elem = 1; return 0; -- 2.1.4

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[alsa-devel] [PATCH v2 0/2] topology: UAPI updates for TLV and ops.
by Liam Girdwood 07 Aug '15

07 Aug '15

This series updates the topology UAPI to allow easier addition of new TLV object types and generic operations for alsa-lib. Mengdong Lin (2): topology: update ABI to improve support for different TLV object types. topology: Add ops support to byte control objects. include/sound/asoc.h | 28 +++++++++++++++++++--------- src/topology/ctl.c | 20 ++++++++------------ 2 files changed, 27 insertions(+), 21 deletions(-) -- 2.1.4

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Re: [alsa-devel] [pulseaudio-discuss] No sound Lubuntu 14.02
by Raymond Yau 07 Aug '15

07 Aug '15

> > >> > >> > > >> > I tried a lot already. No success to get the sound working on my old > maxdata laptop. > >> > This is what I got: > >> > > >> > sl@max:~$ lspci -nnk | grep -iA2 audio > >> > 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW > (ICH6 Family) High Definition Audio Controller [8086:2668] (rev 04) > >> > Subsystem: QUANTA Computer Inc Device [152d:0748] > >> > Kernel driver in use: snd_hda_intel > >> > > >> > sl@max:~$ aplay -l > >> > **** Liste der Hardware-Geräte (PLAYBACK) **** > >> > Karte 0: Intel [HDA Intel], Gerät 6: Si3054 Modem [Si3054 Modem] > >> > Sub-Geräte: 1/1 > >> > Sub-Gerät #0: subdevice #0 > >> > > >> > sl@max:~$ aplay /usr/share/sounds/alsa/Front_Center.wav > >> > Wiedergabe: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed > 16 bit Little Endian, Rate: 48000 Hz, mono > >> > > >> > Please let me know if/what further information I should provide to > give you a better understanding in what is wrong. > >> > > >> > >> > https://lists.fedoraproject.org/pipermail/test/2014-November/123662.html > >> > >> If bios did not setup pin default of alc880 codec, you need to find out > the pin of hp and mic jacks by hdajacksensetest , speaker and internal mic > need to find out by trial and err with hdajackretask > >> > >> _______________________________________________ > >> pulseaudio-discuss mailing list > >> pulseaudio-discuss(a)lists.freedesktop.org > >> http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > > > > This is the output of hda-jack-sense-test. What do you recommend? > > > > root@max:~# hda-jack-sense-test > > Pin 0x14 (Unknown Line Out): present = No > > Pin 0x15 (Unknown Line Out): present = No > > Pin 0x16 (Unknown Line Out): present = No > > Pin 0x17 (Unknown Line Out): present = No > > Pin 0x18 (Unknown Line Out): present = No > > Pin 0x19 (Unknown Line Out): present = No > > Pin 0x1a (Unknown Line Out): present = No > > Pin 0x1b (Unknown Line Out): present = No > > Pin 0x1c (Unknown Line Out): present = No > > Pin 0x1d (Unknown Line Out): present = No > > Pin 0x1e (Unknown Line Out): present = No > > Pin 0x1f (Unknown Line Out): present = No > > > > Post the output of alsa-info.sh > > You have to use > > hdajacksensetest -a > > It gives the same output. > > Did you plug and unplug headphone or mic jack during test ? > > No. > you have to post output of hdajacksensetest -a when you plug headphone , unplug headphone , plug mic and unplug mic > > Since pin default of all pin complex are zero which is Line Out > > It is a laptop. I want to use its internal speakers, if that info helps. > All pins have to be fixed by hdajackretask !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x01000000 0x1f 0x00000000 > Any redundant pins must set to [N/A] > > alc880 only have models for desktop > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/… > > > In the past, there was a "test" model which setup all pins https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… you can try options snd-hda-intel model=6stack-dig to find out whether the speaker is node 0x14 and Headphone and 0x1b you have to manually find front mic, rear mic and line in is your laptop internal mic and mic jacks Do windows driver contain info about the pins since it also need to setup the pin complex too ? sl@max:~$ cat /tmp/alsa-info.txt.5lh8RkGutE upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Fri Aug 7 11:57:07 UTC 2015 !!Linux Distribution !!------------------ Ubuntu 14.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL=" http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: MAXDATA Product Name: PRO8100IS58 Product Version: 5123660003 Firmware Version: M3B61 !!Kernel Information !!------------------ Kernel release: 3.16.0-45-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.16.0-45-generic Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xc0000000 irq 46 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 152d:0748 -- Prefetchable memory behind bridge: 0000000040000000-00000000401fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC880 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0880 Subsystem Id: 0x08800000 Revision Id: 0x100800 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 7 0x18* 0x19 0x1a 0x1b 0x1c 0x14 0x15 Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 7 0x18 0x19 0x1a* 0x1b 0x1c 0x14 0x15 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 10 0x18 0x19 0x1a 0x1b 0x1c* 0x0b 0x14 0x15 0x16 0x17 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0x80 0x80] [0xa3 0xa3] [0xa3 0xa3] Connection: 8 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x11 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x12 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x13 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x19 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x11 Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x12 Node 0x1b [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x13 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="Line Out Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01000000: [Jack] Line Out at Ext Rear Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=25 Unsolicited: tag=00, enabled=0 Connection: 0 Codec: Motorola Si3054 Address: 1 MFG Function Id: 0x2 (unsol 1) Vendor Id: 0x10573055 Subsystem Id: 0x10573055 Revision Id: 0x100700 Modem Function Group: 0x1 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 2 Aug 7 12:03 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 Aug 7 12:03 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 6 Aug 7 12:03 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 4 Aug 7 12:03 /dev/snd/pcmC0D6c crw-rw----+ 1 root audio 116, 3 Aug 7 12:03 /dev/snd/pcmC0D6p crw-rw----+ 1 root audio 116, 1 Aug 7 12:03 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 7 12:03 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Aug 7 12:03 . drwxr-xr-x 3 root root 200 Aug 7 12:03 .. lrwxrwxrwx 1 root root 12 Aug 7 12:03 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xc0000000 irq 46' Mixer name : 'Realtek ALC880' Components : 'HDA:10ec0880,08800000,00100800 HDA:10573055,10573055,00100700' Controls : 8 Simple ctrls : 4 Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 65 Mono: Front Left: Playback 0 [0%] [-35.00dB] [off] Front Right: Playback 0 [0%] [-35.00dB] [off] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!-------------- --startcollapse-- state.Intel { control.1 { iface CARD name 'Line Out Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 65' dbmin -3500 dbmax 3000 dbvalue.0 -3500 dbvalue.1 -3500 } } control.3 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Off-hook Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.5 { iface MIXER name 'Caller ID Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 6 name 'Playback Channel Map' value 0 comment { access read type INTEGER count 1 range '0 - 36' } } control.7 { iface PCM device 6 name 'Capture Channel Map' value 0 comment { access read type INTEGER count 1 range '0 - 36' } } control.8 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x01000000 0x1f 0x00000000 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC0D1/init_pin_configs: /sys/class/sound/hwC0D1/driver_pin_configs: /sys/class/sound/hwC0D1/user_pin_configs: /sys/class/sound/hwC0D1/init_verbs: /sys/class/sound/hwC0D1/hints: !!ALSA/HDA dmesg !!-------------- [ 29.335966] ipw2200: Detected geography ZZE (13 802.11bg channels, 19 802.11a channels) [ 29.378618] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 29.673379] sound hdaudioC0D0: autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 29.673387] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 29.673390] sound hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 29.673393] sound hdaudioC0D0: mono: mono_out=0x1e [ 29.673396] sound hdaudioC0D0: inputs: [ 30.010420] Bluetooth: RFCOMM TTY layer initialized

1 0

[alsa-devel] bug report - headphones stopped working
by Tom Tromey 07 Aug '15

07 Aug '15

A recent kernel upgrade caused my headphones to stop working. I asked on #alsa about this and after a lot of help from "debianuser" (thank you very much), he suggested I report this to this list. I'm using x86-64 Fedora 22. The kernel that works is kernel-4.0.5-300.fc22.x86_64. The kernel that fails is kernel-4.1.3-200.fc22.x86_64. (These are the RPM package names in case that isn't clear.) The output from alsa-info.sh for the working case: http://www.alsa-project.org/db/?f=1da69e358935e2d4a35513fc308a74a07925a4fd ... and for the failing case: http://www.alsa-project.org/db/?f=6a1439affc507f1b83f99b5f42b7cbfe0664f9e8 I'm happy to provide other information as needed, though you'll have to tell me exactly what to do, and turnaround may be on the slow side. thanks, Tom

1 0

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Add DSP management routines
by Vinod Koul 07 Aug '15

07 Aug '15

This is second rev of series to add DSP management routines for SKL driver The first two patches are fixes to use ACPI header and size fix on NHLT query Then we add DSP module configuration calculation helpers followed by SRC and updown calculation. These are used by next patch which adds module creation, and binding routines. Lastly modules are used by pipes so we add pipelines creation, destroy and scheduling helpers Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (5): ASoC: Intel: Skylake: Fix the NHLT rate size ASoC: Intel: Skylake: Use acpi header for NHLT header ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers sound/soc/intel/skylake/skl-messages.c | 751 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-nhlt.h | 14 +- sound/soc/intel/skylake/skl-topology.h | 286 ++++++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 ++++ 5 files changed, 1128 insertions(+), 14 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 1.9.1

2 12

[alsa-devel] [PATCH 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 07 Aug '15

07 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoc: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 ++ arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 3 + sound/soc/rockchip/rockchip_spdif.c | 375 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.4.6

5 10

[alsa-devel] [PATCH 29/31] sound/oxfw: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/firewire/oxfw/oxfw-stream.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 873d40f..77ad5b9 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -512,12 +512,11 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, if (err < 0) goto end; - formats[eid] = kmalloc(*len, GFP_KERNEL); + formats[eid] = kmemdup(buf, *len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; goto end; } - memcpy(formats[eid], buf, *len); /* apply the format for each available sampling rate */ for (i = 0; i < ARRAY_SIZE(oxfw_rate_table); i++) { @@ -531,12 +530,11 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, continue; eid++; - formats[eid] = kmalloc(*len, GFP_KERNEL); + formats[eid] = kmemdup(buf, *len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; goto end; } - memcpy(formats[eid], buf, *len); formats[eid][2] = avc_stream_rate_table[i]; } @@ -594,12 +592,11 @@ static int fill_stream_formats(struct snd_oxfw *oxfw, if (err < 0) break; - formats[eid] = kmalloc(len, GFP_KERNEL); + formats[eid] = kmemdup(buf, len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; break; } - memcpy(formats[eid], buf, len); /* get next entry */ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; -- 1.9.1

2 1

[alsa-devel] [PATCH 30/31] sound/atom: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 5a27861..3dc7358 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -352,10 +352,9 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, * copy from mailbox **/ if (msg_high.part.large) { - data = kzalloc(msg_low, GFP_KERNEL); + data = kmemdup((void *)msg->mailbox_data, msg_low, GFP_KERNEL); if (!data) return; - memcpy(data, (void *) msg->mailbox_data, msg_low); /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; cmd_id = dsp_hdr->cmd_id; -- 1.9.1

2 2

[alsa-devel] [PATCH 31/31] sound/soc: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..a5da91b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -831,12 +831,12 @@ static int soc_tplg_denum_create_values(struct soc_enum *se, if (ec->items > sizeof(*ec->values)) return -EINVAL; - se->dobj.control.dvalues = - kmalloc(ec->items * sizeof(u32), GFP_KERNEL); + se->dobj.control.dvalues = kmemdup(ec->values, + ec->items * sizeof(u32), + GFP_KERNEL); if (!se->dobj.control.dvalues) return -ENOMEM; - memcpy(se->dobj.control.dvalues, ec->values, ec->items * sizeof(u32)); return 0; } -- 1.9.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.2
by Mark Brown 07 Aug '15

07 Aug '15

The following changes since commit 74d33293e467df61de1b1d8b2fbe29e550dec33b: Linux 4.2-rc5 (2015-08-02 18:34:55 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.2-rc5 for you to fetch changes up to d00a9e02178401433fc386e69c936f2039f07b57: Merge remote-tracking branches 'asoc/fix/cs4265', 'asoc/fix/intel' and 'asoc/fix/topology' into asoc-linus (2015-08-06 12:39:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.2 There are a couple of small driver specific fixes here but the overwhelming bulk of these changes are fixes to the topology ABI that has been newly introduced in v4.2. Once this makes it into a release we will have to firm this up but for now getting enhancements in before they've made it into a release is the most expedient thing. ---------------------------------------------------------------- Axel Lin (1): ASoC: cs4265: Fix setting dai format for Left/Right Justified Bard Liao (1): ASoC: rt5645: Fix lost pin setting for DMIC1 Jie Yang (2): ASoC: Intel: haswell: fix initialize 'NULL device *' issue ASoC: Intel: sst_byt: fix initialize 'NULL device *' issue Liam Girdwood (1): ASoC: topology: Add private data type and bump ABI version to 3 Mark Brown (3): Merge tag 'asoc-fix-v4.2-rc3' into asoc-fix-topology Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/cs4265', 'asoc/fix/intel' and 'asoc/fix/topology' into asoc-linus Mengdong Lin (3): ASoC: topology: fix typo in soc_tplg_kcontrol_bind_io() ASoC: topology: Update TLV support so we can support more TLV types ASoC: topology: Add ops support to byte controls UAPI Subhransu S. Prusty (1): ASoC: topology: Add subsequence in topology Vinod Koul (1): ASoC: topology: add private data to manifest include/uapi/sound/asoc.h | 35 +++++++++++------ sound/soc/codecs/cs4265.c | 10 ++--- sound/soc/codecs/rt5645.c | 2 + sound/soc/codecs/rt5645.h | 4 ++ sound/soc/intel/baytrail/sst-baytrail-ipc.c | 2 + sound/soc/intel/haswell/sst-haswell-ipc.c | 2 + sound/soc/soc-topology.c | 61 ++++++++++++++++++++--------- 7 files changed, 82 insertions(+), 34 deletions(-)

2 1

[alsa-devel] [PATCH 00/42] ASoC: Replace users of TLV_DB_RANGE_HEAD
by Lars-Peter Clausen 07 Aug '15

07 Aug '15

Usage of TLV_DB_RANGE_HEAD() is deprecated in favor of DECLARE_TLV_DB_RANGE(). The reason for this is that the former requires to specify the number of items inside the range container while the later is able to calculate this automatically and hence leaves less room for error. While non of the current users of TLV_DB_RANGE_HEAD() gets it wrong it still makes sense to clean this up since existing drivers are often used as templates for new drivers. As an added bonus the code is also slightly shorter. There are also a few drivers which use TLV_DB_RANGE_HEAD() with only a single sub-item. Those are updated to not use a range container at all since it is not necessary. Most of the conversion was auto-generated using the following coccinelle script: // <smpl> @@ declarer name DECLARE_TLV_DB_RANGE; identifier tlv; constant x; @@ -unsigned int tlv[] = { - TLV_DB_RANGE_HEAD(x), +DECLARE_TLV_DB_RANGE(tlv, ... -}; +); // </smpl> - Lars Lars-Peter Clausen (42): ASoC: 88pm860x: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ab8500: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: adau1373: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5623: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5632: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l52: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l56: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l73: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7210: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7213: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da9055: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: jz4740: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9768: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98088: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98095: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9850: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9877: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5631: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5640: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5645: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5651: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5670: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5677: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: sgtl5000: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ssm2602: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: tpa6130a2: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: twl4030: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: uda1380: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8350: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8400: Don't use range container for TLV with one entry ASoC: wm8737: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8753: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8961: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8962: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8990: Don't use range container for TLV with one entry ASoC: wm8991: Don't use range container for TLVs with one entry ASoC: wm8993: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9081: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9713: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm_hubs: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE sound/soc/codecs/88pm860x-codec.c | 28 +++++++++------------ sound/soc/codecs/ab8500-codec.c | 7 +++--- sound/soc/codecs/adau1373.c | 21 +++++++--------- sound/soc/codecs/alc5623.c | 7 +++--- sound/soc/codecs/alc5632.c | 7 +++--- sound/soc/codecs/cs42l52.c | 7 +++--- sound/soc/codecs/cs42l56.c | 21 +++++++--------- sound/soc/codecs/cs42l73.c | 14 +++++------ sound/soc/codecs/da7210.c | 22 +++++++---------- sound/soc/codecs/da7213.c | 15 +++++------ sound/soc/codecs/da9055.c | 15 +++++------ sound/soc/codecs/jz4740.c | 7 +++--- sound/soc/codecs/max9768.c | 7 +++--- sound/soc/codecs/max98088.c | 23 ++++++++--------- sound/soc/codecs/max98090.c | 42 ++++++++++++++----------------- sound/soc/codecs/max98095.c | 35 +++++++++++--------------- sound/soc/codecs/max9850.c | 7 +++--- sound/soc/codecs/max9877.c | 14 +++++------ sound/soc/codecs/rt5631.c | 7 +++--- sound/soc/codecs/rt5640.c | 7 +++--- sound/soc/codecs/rt5645.c | 7 +++--- sound/soc/codecs/rt5651.c | 7 +++--- sound/soc/codecs/rt5670.c | 7 +++--- sound/soc/codecs/rt5677.c | 7 +++--- sound/soc/codecs/sgtl5000.c | 7 +++--- sound/soc/codecs/ssm2602.c | 7 +++--- sound/soc/codecs/tpa6130a2.c | 14 +++++------ sound/soc/codecs/twl4030.c | 7 +++--- sound/soc/codecs/uda1380.c | 14 +++++------ sound/soc/codecs/wm8350.c | 7 +++--- sound/soc/codecs/wm8400.c | 5 +--- sound/soc/codecs/wm8737.c | 7 +++--- sound/soc/codecs/wm8753.c | 7 +++--- sound/soc/codecs/wm8961.c | 7 +++--- sound/soc/codecs/wm8962.c | 14 +++++------ sound/soc/codecs/wm8990.c | 5 +--- sound/soc/codecs/wm8991.c | 52 +++++++-------------------------------- sound/soc/codecs/wm8993.c | 7 +++--- sound/soc/codecs/wm9081.c | 7 +++--- sound/soc/codecs/wm9090.c | 21 +++++++--------- sound/soc/codecs/wm9713.c | 7 +++--- sound/soc/codecs/wm_hubs.c | 7 +++--- 42 files changed, 216 insertions(+), 327 deletions(-) -- 2.1.4

6 91

[alsa-devel] External plugin question
by Mina Almasry 07 Aug '15

07 Aug '15

I'm new to alsa dev and I'm trying to write a very simple external pcm plugin. I want my pcm plugin to output sound to multiple slave pcms when sound is being writei'd to it. So: snd_pcm_myplugin_writei(pcm, buffer, size) { foreach slave pcm { snd_pcm_writei(slave_pcm, buffer, size); } } I'm having trouble implementing this plugin because the external plugin sdk does not allow me to define the writei function for my plugin, but lets me define the transfer function which has a very different signature than writei: transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) how do I extract buffer and size from these arguments? Any tips on how I write my plugin. Mina

2 1

[alsa-devel] External plugin question
by Mina Almasry 07 Aug '15

07 Aug '15

I'm new to alsa dev and I'm trying to write a very simple external pcm plugin. I want my pcm plugin to output sound to multiple slave pcms when sound is being writei'd to it. So: snd_pcm_myplugin_writei(pcm, buffer, size) { foreach slave pcm { snd_pcm_writei(slave_pcm, buffer, size); } } I'm having trouble implementing this plugin because the external plugin sdk does not allow me to define the writei function for my plugin, but lets me define the transfer function which has a very different signature than writei: transfer(snd_pcm_ioplug_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___i_o_plug.html…> *io, const snd_pcm_channel_area_t <http://www.alsa-project.org/alsa-doc/alsa-lib/structsnd__pcm__channel__area…> *areas, snd_pcm_uframes_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gab01fcfe9…> offset, snd_pcm_uframes_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gab01fcfe9…> size) how do I extract buffer and size from these arguments? Any tips on how I write my plugin. (I'm aware of the multi plugin but I don't wont to use that. I want to write my own and introduce custom logic later.) Mina

1 0

[alsa-devel] [PATCH 2/2] ASoC: max98090: Enforce correct device sequencing when configuring a new
by yang.a.fang＠intel.com 07 Aug '15

07 Aug '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> audio route when there is an existing active audio route(s). This patch fixed recording noise issue while playback is active. We have some registers which require the device to be in full shutdown or to enter full shutdown before the register settings will take effect. Currently the driver is not shutting down the device when a new audio route is created. If a new audio route is made active while there is already an active audio route, then the required register sequencing is violated. A hardware shutdown toggle when creating a new audio route corrects the sequencing error. The device must remain in hardware shutdown for 40ms to allow the internal hardware core to fully shutdown. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/codecs/max98090.c | 46 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/max98090.h | 1 + 2 files changed, 41 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 91c9f9f..1516c6b 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -850,6 +850,19 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, return 0; } +static int max98090_shdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (event & SND_SOC_DAPM_POST_PMU) + max98090->shdn_pending = TRUE; + + return 0; + +} + static const char *mic1_mux_text[] = { "IN12", "IN56" }; static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, @@ -1158,9 +1171,11 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, M98090_SDOEN_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICL_SHIFT, 0, NULL, 0), + M98090_DIGMICL_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICR_SHIFT, 0, NULL, 0), + M98090_DIGMICR_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, M98090_AHPF_SHIFT, 0, NULL, 0), @@ -1205,10 +1220,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_adc_mixer_controls[0], ARRAY_SIZE(max98090_right_adc_mixer_controls)), - SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADLEN_SHIFT, 0), - SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADREN_SHIFT, 0), + SND_SOC_DAPM_ADC_E("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), @@ -2539,9 +2556,26 @@ static int max98090_remove(struct snd_soc_codec *codec) return 0; } +static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq, bool power_up) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (max98090->shdn_pending && power_up) { + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(40); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + max98090->shdn_pending = FALSE; + } +} + static struct snd_soc_codec_driver soc_codec_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, + .seq_notifier = max98090_seq_notifier, .set_bias_level = max98090_set_bias_level, }; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 21ff743..bc610d9 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1543,6 +1543,7 @@ struct max98090_priv { unsigned int pa2en; unsigned int sidetone; bool master; + bool shdn_pending; }; int max98090_mic_detect(struct snd_soc_codec *codec, -- 1.7.9.5

1 0

[alsa-devel] [asoc:fix/topology 5/7] sound/soc/soc-topology.c:596:14: sparse: incorrect type in assignment (different base types)
by kbuild test robot 06 Aug '15

06 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git fix/topology head: c7bcf8777a539e64dafc7417c00047aee6eb8909 commit: 28a87eebcad40101b1b273cbd4f2a02c104f9367 [5/7] ASoC: topology: Update TLV support so we can support more TLV types reproduce: # apt-get install sparse git checkout 28a87eebcad40101b1b273cbd4f2a02c104f9367 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) sound/soc/soc-topology.c:1696:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:190:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:191:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:191:39: expected int sound/soc/soc-topology.c:191:39: got restricted __le32 [usertype] reg sound/soc/soc-topology.c:203:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:204:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:204:39: expected int sound/soc/soc-topology.c:204:39: got restricted __le32 [usertype] shift sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:519:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:521:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:523:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:534:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:536:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:538:62: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:596:14: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:596:14: expected unsigned int [unsigned] <noident> sound/soc/soc-topology.c:596:14: got restricted __le32 [usertype] min sound/soc/soc-topology.c:597:22: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:609:17: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:612:15: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:616:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:667:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:677:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:677:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:677:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:679:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:679:26: expected int [signed] max sound/soc/soc-topology.c:679:26: got restricted __le32 [usertype] max sound/soc/soc-topology.c:748:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:758:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:758:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:758:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:770:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:770:25: expected int [signed] max sound/soc/soc-topology.c:770:25: got restricted __le32 [usertype] max sound/soc/soc-topology.c:771:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:771:25: expected int [signed] min sound/soc/soc-topology.c:771:25: got restricted __le32 [usertype] min sound/soc/soc-topology.c:772:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:772:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:772:28: got restricted __le32 [usertype] invert sound/soc/soc-topology.c:773:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:773:34: expected int [signed] platform_max sound/soc/soc-topology.c:773:34: got restricted __le32 [usertype] platform_max sound/soc/soc-topology.c:824:44: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:828:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:855:15: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:859:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:863:9: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:887:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:905:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:905:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:905:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:913:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:913:27: expected unsigned int [unsigned] items sound/soc/soc-topology.c:913:27: got restricted __le32 [usertype] items sound/soc/soc-topology.c:914:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:914:26: expected unsigned int [unsigned] mask sound/soc/soc-topology.c:914:26: got restricted __le32 [usertype] mask sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:994:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:994:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1001:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1013:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1013:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1013:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1020:59: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1020:59: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1020:59: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1023:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1023:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1023:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1040:24: sparse: incorrect type in initializer (different base types) sound/soc/soc-topology.c:1040:24: expected int [signed] count sound/soc/soc-topology.c:1040:24: got restricted __le32 [usertype] count sound/soc/soc-topology.c:1043:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1043:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1049:27: sparse: incorrect type in argument 4 (different base types) sound/soc/soc-topology.c:1049:27: expected unsigned long [unsigned] [usertype] bytes sound/soc/soc-topology.c:1049:27: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1107:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1120:30: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1120:30: expected unsigned int [unsigned] access sound/soc/soc-topology.c:1120:30: got restricted __le32 [usertype] access sound/soc/soc-topology.c:1132:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1132:25: expected int [signed] max sound/soc/soc-topology.c:1132:25: got restricted __le32 [usertype] max sound/soc/soc-topology.c:1133:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1133:25: expected int [signed] min sound/soc/soc-topology.c:1133:25: got restricted __le32 [usertype] min sound/soc/soc-topology.c:1134:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1134:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:1134:28: got restricted __le32 [usertype] invert sound/soc/soc-topology.c:1135:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1135:34: expected int [signed] platform_max sound/soc/soc-topology.c:1135:34: got restricted __le32 [usertype] platform_max sound/soc/soc-topology.c:1179:25: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1200:20: sparse: incorrect type in assignment (different base types) vim +596 sound/soc/soc-topology.c 580 return 0; 581 } 582 583 584 static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, 585 struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) 586 { 587 unsigned int item_len = 2 * sizeof(unsigned int); 588 unsigned int *p; 589 590 p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); 591 if (!p) 592 return -ENOMEM; 593 594 p[0] = SNDRV_CTL_TLVT_DB_SCALE; 595 p[1] = item_len; > 596 p[2] = scale->min; 597 p[3] = (scale->step & TLV_DB_SCALE_MASK) 598 | (scale->mute ? TLV_DB_SCALE_MUTE : 0); 599 600 kc->tlv.p = (void *)p; 601 return 0; 602 } 603 604 static int soc_tplg_create_tlv(struct soc_tplg *tplg, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8731: Drop wm8731_writeable function
by Axel Lin 06 Aug '15

06 Aug '15

When .max_register is set and .writeable_reg is not implement, registers between 0 and .max_register are writeable. This is the same as current implementation of wm8731_writeable(), so just drop implementation for .writeable_reg callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8731.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c862232..ace8645 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -79,11 +79,6 @@ static bool wm8731_volatile(struct device *dev, unsigned int reg) return reg == WM8731_RESET; } -static bool wm8731_writeable(struct device *dev, unsigned int reg) -{ - return reg <= WM8731_RESET; -} - #define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -655,7 +650,6 @@ static const struct regmap_config wm8731_regmap = { .max_register = WM8731_RESET, .volatile_reg = wm8731_volatile, - .writeable_reg = wm8731_writeable, .cache_type = REGCACHE_RBTREE, .reg_defaults = wm8731_reg_defaults, -- 2.1.0

3 5

[alsa-devel] [PATCH] ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation
by Axel Lin 06 Aug '15

06 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 430fa7d..9723e44 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -278,51 +278,38 @@ static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (freq) { case 0: wm8741->sysclk_constraints = NULL; - wm8741->sysclk = freq; - return 0; - + break; case 11289600: wm8741->sysclk_constraints = &constraints_11289; - wm8741->sysclk = freq; - return 0; - + break; case 12288000: wm8741->sysclk_constraints = &constraints_12288; - wm8741->sysclk = freq; - return 0; - + break; case 16384000: wm8741->sysclk_constraints = &constraints_16384; - wm8741->sysclk = freq; - return 0; - + break; case 16934400: wm8741->sysclk_constraints = &constraints_16934; - wm8741->sysclk = freq; - return 0; - + break; case 18432000: wm8741->sysclk_constraints = &constraints_18432; - wm8741->sysclk = freq; - return 0; - + break; case 22579200: case 33868800: wm8741->sysclk_constraints = &constraints_22579; - wm8741->sysclk = freq; - return 0; - + break; case 24576000: wm8741->sysclk_constraints = &constraints_24576; - wm8741->sysclk = freq; - return 0; - + break; case 36864000: wm8741->sysclk_constraints = &constraints_36864; - wm8741->sysclk = freq; - return 0; + break; + default: + return -EINVAL; } - return -EINVAL; + + wm8741->sysclk = freq; + return 0; } static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 2.1.0

3 2

[alsa-devel] query on upstream ASoC driver split across SOUND and DRM subsystems
by Bayyavarapu, Maruthi 06 Aug '15

06 Aug '15

Hi, I want to send AMD ASoC driver for review. This driver has a dependency with AMD's 'DRM' subsystem components, which are not upstreamed yet. Please suggest, whether this ASoC driver can be upstreamed via 'DRM' tree or any alternate approach is preferred. The link to ASoC driver is : http://cgit.freedesktop.org/~agd5f/linux/commit/?h=acp-wip&id=2b67a44150988… ASoC commits and dependencies are currently in 'acp-wip' branch of : git://people.freedesktop.org/~agd5f/linux Thanks, Maruthi

3 2

[alsa-devel] [PATCH v2 0/3] ASoC: topology: UAPI updates for TLV, ops and types
by Liam Girdwood 06 Aug '15

06 Aug '15

This series updates the topology UAPI to allow easier addition of new TLV object types, generic operations and added a private data type for the kernel topology core. Changes since V1 * Split into several patches for UAPI changes. * Fixed {} usage in if statement. * Added __le32 data[SND_SOC_TPLG_TLV_SIZE] back to struct snd_soc_tplg_ctl_tlv Liam Girdwood (1): ASoC: topology: Add private data type and bump ABI version to 3 Mengdong Lin (2): ASoC: topology: Update TLV support so we can support more TLV types ASoC: topology: Add ops support to byte controls UAPI include/uapi/sound/asoc.h | 33 ++++++++++++++++++--------- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 63 insertions(+), 28 deletions(-) -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: rl6231: Simplify DMIC divider calculation expression
by Anatol Pomozov 06 Aug '15

06 Aug '15

Existing implementation checks all divider values and tracks 'red' proximity value for the frequency. But as divider array is monotonically increasing the first divider that gives DMIC rate in 3MHz range is the best one we should use. No need for 'red' zone tracking. Additionally make sure that DMIC frequency is higher 1MHz. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rl6231.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 57e51c1..aca479f 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -62,31 +62,31 @@ int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft) EXPORT_SYMBOL_GPL(rl6231_get_pre_div); /** - * rl6231_calc_dmic_clk - Calculate the parameter of dmic. + * rl6231_calc_dmic_clk - Calculate the frequency divider parameter of dmic. * * @rate: base clock rate. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. + * Choose divider parameter that gives the highest possible DMIC frequency in + * 1MHz - 3MHz range. */ int rl6231_calc_dmic_clk(int rate) { - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; - int i, red, bound, temp; + int div[] = {2, 3, 4, 6, 8, 12}; + int i; + + if (rate < 1000000 * div[0]) { + pr_warn("Base clock rate %d is too low\n", rate); + return -EINVAL; + } - red = 3000000 * 12; for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } + /* find divider that gives DMIC frequency below 3MHz */ + if (3000000 * div[i] >= rate) + return i; } - return idx; + pr_warn("Base clock rate %d is too high\n", rate); + return -EINVAL; } EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); -- 2.5.0.rc2.392.g76e840b

3 2

[alsa-devel] [PATCH v2] ASoC: topology: Use kcalloc instead of kzalloc for array allocation
by Axel Lin 06 Aug '15

06 Aug '15

Also remove unnecessary memset. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: One more kcalloc conversion in soc_tplg_dapm_widget_dmixer_create sound/soc/soc-topology.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..e4b7fc6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1069,7 +1069,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( struct snd_soc_tplg_mixer_control *mc; int i, err; - kc = kzalloc(sizeof(*kc) * num_kcontrols, GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; @@ -1250,7 +1250,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( struct snd_kcontrol_new *kc; int i, err; - kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + kc = kcalloc(count, sizeof(*kc), GFP_KERNEL); if (!kc) return NULL; @@ -1273,7 +1273,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - memset(kc, 0, sizeof(*kc)); kc[i].name = be->hdr.name; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; -- 2.1.0

2 1

[alsa-devel] Verb for 0x17 pin config
by Kailang 06 Aug '15

06 Aug '15

Hi Takashi, commit http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=977e… ALSA: hda/realtek - Support Dell headset mode for ALC298 Dell had use verb to do speaker preset. So, The same SSID maybe 0x17 verb was different. 0x17 0x90170110 0x17 0x90170140 0x17 0x90170150 Upper verb will show on this case. Could I change it to quirk for SSID? BR, Kailang

2 6

Re: [alsa-devel] [ALC668]: Asus N751JK - mic selection not working
by Raymond Yau 06 Aug '15

06 Aug '15

>> >> Ports: >> >> analog-output-speaker: Speakers (priority: 10000) >> analog-output-headphones: Headphones (priority: 9000, not available) >> Active Port: analog-output-speaker >> >> >> When you select model=dell-headset-multi, the driver create "Headphone Mic Jack" kctl with type 1 which is SND_JACK_HEADPHONE >> >> type 2 is SND_JACK_MICROPHONE >> >> type 3 is SND_JACK_HEADSET but hda jack does not use this type >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/s… >> >> JACK created Headphone Mic, type 1 >> CTRL: add: Headphone Mic Jack:0 >> JACK report Headphone Mic, status 0 >> JACK created Headset Mic Phantom, type 2 >> CTRL: add: Headset Mic Phantom Jack:0 >> JACK report Headset Mic Phantom, status 2 >> CTL Notify: Headset Mic Phantom Jack:0, mask=1 >> JACK created Internal Mic Phantom, type 2 >> CTRL: add: Internal Mic Phantom Jack:0 >> JACK report Internal Mic Phantom, status 2 >> CTL Notify: Internal Mic Phantom Jack:0, mask=1 >> JACK created Speaker Front Phantom, type 4 >> CTRL: add: Speaker Front Phantom Jack:0 >> JACK report Speaker Front Phantom, status 4 >> CTL Notify: Speaker Front Phantom Jack:0, mask=1 >> JACK created External Subwoofer, type 4 >> CTRL: add: External Subwoofer Jack:0 >> >> The problem of hda jack kctls are those retaskable I/O jack controls >> >> 1) multi io jacks in those 3 stack desktop >> 2) headphone/mic jack in asus netbook >> 3) dell-headset-multi combo jack >> >> the role of the jack in 1) change from input to output when "Channel mode" change from "2ch" to "4ch" or "6ch" >> >> the role of the jack in 2) and 3) change from Output to Input when "Capture Source" change from Internal Mic/Headset Mic to Headphone Mic >> >> In your case, Headphone Mic Jack kctl return the presence of HP jack except when the combo jack is set to support conventional mic (i.e. when the driver change pin ctl of hp pin node 0x15 to zero and set pin ctl of mic pin to IN), >> >> The driver already unmute the internal speaker when you change capture source to Headphone Mic >> >> The only point is whether pulseaudio still regard this Headphone Mic Jack as the availability of the headphone port when the capture source is Headphone Mic >> >> http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ca4942e89cf462… >> >> > > > I really appreciate your efforts in this matter. Unfortunately I understand parts of our talks and I am unsure what I can do. > I am not a kernel / driver developer. > > Are there any steps I can do knowing my limited skills on driver development? > If so, would you mind to describe them step by step, so this beginner understands? :D Don't forget cc David since he implement dell-headset-multi https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} You have to post the output of pactl list sources when the is no jack plugged, headphone plugged, headset plugged and mic jack plugged The logic is defined in alc_update_headset_mode() Although auto mic selection is disabled when the combo jack can support conventional mic and headset mic The capture source of these kind of laptop is not as same as the capture source of ac97 codec of pci sound cards, it is not really manually selected in sound preference When there is no jack plugged in, the capture source should be internal mic. As the codec cannot differentitate headset, headphone and mic when there is jack plugged in and driver use mux_pin to determine the type of the jack In pulseaudio term, [Jack Headphone Mic] can determine the availability of internal mic is YES when unplugged but unknown when plugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headphone mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headset mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… It is not exactly UNKNOWN but determined by the capture source when plugged

2 4

[alsa-devel] [PATCH 5/6][RESEND] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 154bcf6..8e5ff5e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

1 0

[alsa-devel] [PATCH 4/6][RESEND] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7f4f0b9..154bcf6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

1 0

[alsa-devel] [PATCH 3/6][RESEND] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a83b900..7f4f0b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

1 0

[alsa-devel] [PATCH 2/6][RESEND] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8185edc..a83b900 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

1 0

[alsa-devel] [PATCH 1/6][RESEND] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 484ff20..8185edc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

1 0

[alsa-devel] [PATCH] ALSA: aoa-soundbus: Switch to dev_pm_ops
by Lars-Peter Clausen 05 Aug '15

05 Aug '15

Update the aoa-soundbus framework to use dev_pm_ops rather than the deprecated legacy suspend and resume callbacks. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single aoa-soundbus driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/aoa/fabrics/layout.c | 21 ++++++++++++--------- sound/aoa/soundbus/core.c | 28 ---------------------------- sound/aoa/soundbus/soundbus.h | 2 -- 3 files changed, 12 insertions(+), 39 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 9dc5806..8f71f7e 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1120,10 +1120,10 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) return 0; } -#ifdef CONFIG_PM -static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state) +#ifdef CONFIG_PM_SLEEP +static int aoa_fabric_layout_suspend(struct device *dev) { - struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); + struct layout_dev *ldev = dev_get_drvdata(dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_off(&ldev->gpio); @@ -1131,15 +1131,19 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta return 0; } -static int aoa_fabric_layout_resume(struct soundbus_dev *sdev) +static int aoa_fabric_layout_resume(struct device *dev) { - struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); + struct layout_dev *ldev = dev_get_drvdata(dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_restore) ldev->gpio.methods->all_amps_restore(&ldev->gpio); return 0; } + +static SIMPLE_DEV_PM_OPS(aoa_fabric_layout_pm_ops, + aoa_fabric_layout_suspend, aoa_fabric_layout_resume); + #endif static struct soundbus_driver aoa_soundbus_driver = { @@ -1147,12 +1151,11 @@ static struct soundbus_driver aoa_soundbus_driver = { .owner = THIS_MODULE, .probe = aoa_fabric_layout_probe, .remove = aoa_fabric_layout_remove, -#ifdef CONFIG_PM - .suspend = aoa_fabric_layout_suspend, - .resume = aoa_fabric_layout_resume, -#endif .driver = { .owner = THIS_MODULE, +#ifdef CONFIG_PM_SLEEP + .pm = &aoa_fabric_layout_pm_ops, +#endif } }; diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 3edf736..70bcaa7 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -126,30 +126,6 @@ static void soundbus_device_shutdown(struct device *dev) drv->shutdown(soundbus_dev); } -#ifdef CONFIG_PM - -static int soundbus_device_suspend(struct device *dev, pm_message_t state) -{ - struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); - struct soundbus_driver * drv = to_soundbus_driver(dev->driver); - - if (dev->driver && drv->suspend) - return drv->suspend(soundbus_dev, state); - return 0; -} - -static int soundbus_device_resume(struct device * dev) -{ - struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); - struct soundbus_driver * drv = to_soundbus_driver(dev->driver); - - if (dev->driver && drv->resume) - return drv->resume(soundbus_dev); - return 0; -} - -#endif /* CONFIG_PM */ - /* soundbus_dev_attrs is declared in sysfs.c */ ATTRIBUTE_GROUPS(soundbus_dev); static struct bus_type soundbus_bus_type = { @@ -158,10 +134,6 @@ static struct bus_type soundbus_bus_type = { .uevent = soundbus_uevent, .remove = soundbus_device_remove, .shutdown = soundbus_device_shutdown, -#ifdef CONFIG_PM - .suspend = soundbus_device_suspend, - .resume = soundbus_device_resume, -#endif .dev_groups = soundbus_dev_groups, }; diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 21e756c..ae40224 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -188,8 +188,6 @@ struct soundbus_driver { int (*probe)(struct soundbus_dev* dev); int (*remove)(struct soundbus_dev* dev); - int (*suspend)(struct soundbus_dev* dev, pm_message_t state); - int (*resume)(struct soundbus_dev* dev); int (*shutdown)(struct soundbus_dev* dev); struct device_driver driver; -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: Use devm_clk_get
by Vaishali Thakkar 05 Aug '15

05 Aug '15

This patch introduces the use of managed resource function devm_clk_get instead of clk_get and removes corresponding calls to clk_put in the probe and remove functions. To be compatible with the change various gotos are replaced with direct returns, and unneeded labels are dropped. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra30_ahub.c | 49 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index bc94e5d..989b1e8 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -549,54 +549,51 @@ static int tegra30_ahub_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Can't get reset %s\n", configlink_mods[i].rst_name); ret = PTR_ERR(rst); - goto err; + return ret; } ret = reset_control_deassert(rst); reset_control_put(rst); if (ret) - goto err; + return ret; } ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), GFP_KERNEL); if (!ahub) { dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } dev_set_drvdata(&pdev->dev, ahub); ahub->soc_data = soc_data; ahub->dev = &pdev->dev; - ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + ahub->clk_d_audio = devm_clk_get(&pdev->dev, "d_audio"); if (IS_ERR(ahub->clk_d_audio)) { dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); ret = PTR_ERR(ahub->clk_d_audio); - goto err; + return ret; } - ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + ahub->clk_apbif = devm_clk_get(&pdev->dev, "apbif"); if (IS_ERR(ahub->clk_apbif)) { dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); ret = PTR_ERR(ahub->clk_apbif); - goto err_clk_put_d_audio; + return ret; } res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res0) { dev_err(&pdev->dev, "No apbif memory resource\n"); - ret = -ENODEV; - goto err_clk_put_apbif; + return -ENODEV; } region = devm_request_mem_region(&pdev->dev, res0->start, resource_size(res0), DRV_NAME); if (!region) { dev_err(&pdev->dev, "request region apbif failed\n"); - ret = -EBUSY; - goto err_clk_put_apbif; + return -EBUSY; } ahub->apbif_addr = res0->start; @@ -604,8 +601,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) resource_size(res0)); if (!regs_apbif) { dev_err(&pdev->dev, "ioremap apbif failed\n"); - ret = -ENOMEM; - goto err_clk_put_apbif; + return -ENOMEM; } ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, @@ -613,31 +609,28 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (IS_ERR(ahub->regmap_apbif)) { dev_err(&pdev->dev, "apbif regmap init failed\n"); ret = PTR_ERR(ahub->regmap_apbif); - goto err_clk_put_apbif; + return ret; } regcache_cache_only(ahub->regmap_apbif, true); res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); if (!res1) { dev_err(&pdev->dev, "No ahub memory resource\n"); - ret = -ENODEV; - goto err_clk_put_apbif; + return -ENODEV; } region = devm_request_mem_region(&pdev->dev, res1->start, resource_size(res1), DRV_NAME); if (!region) { dev_err(&pdev->dev, "request region ahub failed\n"); - ret = -EBUSY; - goto err_clk_put_apbif; + return -EBUSY; } regs_ahub = devm_ioremap(&pdev->dev, res1->start, resource_size(res1)); if (!regs_ahub) { dev_err(&pdev->dev, "ioremap ahub failed\n"); - ret = -ENOMEM; - goto err_clk_put_apbif; + return -ENOMEM; } ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, @@ -645,7 +638,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (IS_ERR(ahub->regmap_ahub)) { dev_err(&pdev->dev, "ahub regmap init failed\n"); ret = PTR_ERR(ahub->regmap_ahub); - goto err_clk_put_apbif; + return ret; } regcache_cache_only(ahub->regmap_ahub, true); @@ -662,12 +655,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) err_pm_disable: pm_runtime_disable(&pdev->dev); -err_clk_put_apbif: - clk_put(ahub->clk_apbif); -err_clk_put_d_audio: - clk_put(ahub->clk_d_audio); - ahub = NULL; -err: + return ret; } @@ -680,11 +668,6 @@ static int tegra30_ahub_remove(struct platform_device *pdev) if (!pm_runtime_status_suspended(&pdev->dev)) tegra30_ahub_runtime_suspend(&pdev->dev); - clk_put(ahub->clk_apbif); - clk_put(ahub->clk_d_audio); - - ahub = NULL; - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 05 Aug '15

05 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoc: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 381 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 536 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.5.0

2 4

[alsa-devel] [PATCH] ASoC: topology: Update ABI with improvements for TLV, operations and private data.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This is the kernel patch - a corresponding alsa-lib patch has also been sent. This patch updates the topology ABI version and adds support for private data types, specfic structures for each TLV data type and refactors the operations mapping structure to be generic (so that it can be used by other types). This code is only used within Intel atm and is not in an official alsa-lib release so impact should be minimali for the ABI update. Currently the TLV topology structure is targeted at only supporting the DB scale data. This patch extends support for the other TLV types so they can be easily added at a later stage. TLV structure is moved to common topology control header since it's a common field for controls and can be processed in a general way. Users must set a proper access flag for a control since it's used to decide if the TLV field is valid and if a TLV callback is needed. Removed the following fields from topology TLV struct: - size/count: type can decide the size. - numid: not needed to initialize TLV for kcontrol. - data: replaced by the type specific struct. Added TLV structure to generic control header and removed TLV structure from mixer control. Added generic ops to geneic control header. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/uapi/sound/asoc.h | 32 +++++++++++++++++--------- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 62 insertions(+), 28 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 2819fc1..69383b0 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -77,7 +77,7 @@ #define SND_SOC_TPLG_NUM_TEXTS 16 /* ABI version */ -#define SND_SOC_TPLG_ABI_VERSION 0x2 +#define SND_SOC_TPLG_ABI_VERSION 0x3 /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 @@ -97,7 +97,8 @@ #define SND_SOC_TPLG_TYPE_PCM 7 #define SND_SOC_TPLG_TYPE_MANIFEST 8 #define SND_SOC_TPLG_TYPE_CODEC_LINK 9 -#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK +#define SND_SOC_TPLG_TYPE_PDATA 10 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA /* vendor block IDs - please add new vendor types to end */ #define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 @@ -137,11 +138,18 @@ struct snd_soc_tplg_private { /* * Kcontrol TLV data. */ +struct snd_soc_tplg_tlv_dbscale { + __le32 min; + __le32 step; + __le32 mute; +} __attribute__((packed)); + struct snd_soc_tplg_ctl_tlv { - __le32 size; /* in bytes aligned to 4 */ - __le32 numid; /* control element numeric identification */ - __le32 count; /* number of elem in data array */ - __le32 data[SND_SOC_TPLG_TLV_SIZE]; + __le32 size; /* in bytes of this structure */ + __le32 type; /* SNDRV_CTL_TLVT_*, type of TLV */ + union { + struct snd_soc_tplg_tlv_dbscale scale; + }; } __attribute__((packed)); /* @@ -155,9 +163,11 @@ struct snd_soc_tplg_channel { } __attribute__((packed)); /* - * Kcontrol Operations IDs + * Genericl Operations IDs, for binding Kcontrol or Bytes ext ops + * Kcontrol ops need get/put/info. + * Bytes ext ops need get/put. */ -struct snd_soc_tplg_kcontrol_ops_id { +struct snd_soc_tplg_io_ops { __le32 get; __le32 put; __le32 info; @@ -171,8 +181,8 @@ struct snd_soc_tplg_ctl_hdr { __le32 type; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 access; - struct snd_soc_tplg_kcontrol_ops_id ops; - __le32 tlv_size; /* non zero means control has TLV data */ + struct snd_soc_tplg_io_ops ops; + struct snd_soc_tplg_ctl_tlv tlv; } __attribute__((packed)); /* @@ -260,7 +270,6 @@ struct snd_soc_tplg_mixer_control { __le32 invert; __le32 num_channels; struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; - struct snd_soc_tplg_ctl_tlv tlv; struct snd_soc_tplg_private priv; } __attribute__((packed)); @@ -304,6 +313,7 @@ struct snd_soc_tplg_bytes_control { __le32 mask; __le32 base; __le32 num_regs; + struct snd_soc_tplg_io_ops ext_ops; struct snd_soc_tplg_private priv; } __attribute__((packed)); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..bebd6f3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-topology.h> +#include <sound/tlv.h> /* * We make several passes over the data (since it wont necessarily be ordered) @@ -579,28 +580,51 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, return 0; } + +static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) +{ + unsigned int item_len = 2 * sizeof(unsigned int); + unsigned int *p; + + p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); + if (!p) + return -ENOMEM; + + p[0] = SNDRV_CTL_TLVT_DB_SCALE; + p[1] = item_len; + p[2] = scale->min; + p[3] = (scale->step & TLV_DB_SCALE_MASK) + | (scale->mute ? TLV_DB_SCALE_MUTE : 0); + + kc->tlv.p = (void *)p; + return 0; +} + static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_hdr *tc) { - struct snd_ctl_tlv *tlv; - int size; + struct snd_soc_tplg_ctl_tlv *tplg_tlv; - if (tplg_tlv->count == 0) + if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) return 0; - size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & - ~(sizeof(unsigned int) - 1)); - tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); - if (tlv == NULL) - return -ENOMEM; - - dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, size); + if (tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) + kc->tlv.c = snd_soc_bytes_tlv_callback; + else { + tplg_tlv = &tc->tlv; + switch (tplg_tlv->type) { + case SNDRV_CTL_TLVT_DB_SCALE: + return soc_tplg_create_tlv_db_scale(tplg, kc, + &tplg_tlv->scale); - tlv->numid = tplg_tlv->numid; - tlv->length = size; - memcpy(&tlv->tlv[0], tplg_tlv->data, size); - kc->tlv.p = (void *)tlv; + /* TODO: add support for other TLV types */ + default: + dev_dbg(tplg->dev, "Unsupported TLV type %d\n", + tplg_tlv->type); + return -EINVAL; + } + } return 0; } @@ -772,7 +796,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->tlv); + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- 2.1.4

2 2

[alsa-devel] [PATCH] ALSA: hda - one Dell machine needs the headphone white noise fixup
by Hui Wang 05 Aug '15

05 Aug '15

The fixup ALC292_FIXUP_DISABLE_AAMIX can fix the white noise of the headphone on this Dell machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b20f104..dd7274d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5189,6 +5189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.9.1

2 1

[alsa-devel] [PATCH] topology: Update ABI to improve TLV, operations and private data support.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This is the alsa-lib patch - a corresponding kernel patch has also been sent. This patch updates the topology ABI version and adds support for private data types, specfic structures for each TLV data type and refactors the operations mapping structure to be generic (so that it can be used by other types). This code is only used within Intel atm and is not in an official alsa-lib release so impact should be minimal for the ABI update. Currently the TLV topology structure is targeted at only supporting the DB scale data. This patch extends support for the other TLV types so they can be easily added at a later stage. TLV structure is moved to common topology control header since it's a common field for controls and can be processed in a general way. Users must set a proper access flag for a control since it's used to decide if the TLV field is valid and if a TLV callback is needed. Removed the following fields from topology TLV struct: - size/count: type can decide the size. - numid: not needed to initialize TLV for kcontrol. - data: replaced by the type specific struct. Added TLV structure to generic control header and removed TLV structure from mixer control. Added generic ops to geneic control header. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/asoc.h | 27 ++++++++++++++++++--------- src/topology/ctl.c | 20 ++++++++------------ 2 files changed, 26 insertions(+), 21 deletions(-) diff --git a/include/sound/asoc.h b/include/sound/asoc.h index bb6dcf3..61b8473 100644 --- a/include/sound/asoc.h +++ b/include/sound/asoc.h @@ -135,11 +135,18 @@ struct snd_soc_tplg_private { /* * Kcontrol TLV data. */ +struct snd_soc_tplg_tlv_dbscale { + __le32 min; + __le32 step; + __le32 mute; +} __attribute__((packed)); + struct snd_soc_tplg_ctl_tlv { - __le32 size; /* in bytes aligned to 4 */ - __le32 numid; /* control element numeric identification */ - __le32 count; /* number of elem in data array */ - __le32 data[SND_SOC_TPLG_TLV_SIZE]; + __le32 size; /* in bytes of this structure */ + __le32 type; /* SNDRV_CTL_TLVT_*, type of TLV */ + union { + struct snd_soc_tplg_tlv_dbscale scale; + }; } __attribute__((packed)); /* @@ -153,9 +160,11 @@ struct snd_soc_tplg_channel { } __attribute__((packed)); /* - * Kcontrol Operations IDs + * Genericl Operations IDs, for binding Kcontrol or Bytes ext ops + * Kcontrol ops need get/put/info. + * Bytes ext ops need get/put. */ -struct snd_soc_tplg_kcontrol_ops_id { +struct snd_soc_tplg_io_ops { __le32 get; __le32 put; __le32 info; @@ -169,8 +178,8 @@ struct snd_soc_tplg_ctl_hdr { __le32 type; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 access; - struct snd_soc_tplg_kcontrol_ops_id ops; - __le32 tlv_size; /* non zero means control has TLV data */ + struct snd_soc_tplg_io_ops ops; + struct snd_soc_tplg_ctl_tlv tlv; } __attribute__((packed)); /* @@ -258,7 +267,6 @@ struct snd_soc_tplg_mixer_control { __le32 invert; __le32 num_channels; struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; - struct snd_soc_tplg_ctl_tlv tlv; struct snd_soc_tplg_private priv; } __attribute__((packed)); @@ -302,6 +310,7 @@ struct snd_soc_tplg_bytes_control { __le32 mask; __le32 base; __le32 num_regs; + struct snd_soc_tplg_io_ops ext_ops; struct snd_soc_tplg_private priv; } __attribute__((packed)); diff --git a/src/topology/ctl.c b/src/topology/ctl.c index aa06ff6..930b508 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -28,10 +28,7 @@ static int copy_tlv(struct tplg_elem *elem, struct tplg_elem *ref) tplg_dbg("TLV '%s' used by '%s\n", ref->id, elem->id); /* TLV has a fixed size */ - mixer_ctrl->tlv = *tlv; - - /* set size of TLV data */ - mixer_ctrl->hdr.tlv_size = tlv->count * sizeof(uint32_t); + mixer_ctrl->hdr.tlv = *tlv; return 0; } @@ -209,20 +206,19 @@ static int tplg_parse_tlv_dbscale(snd_config_t *cfg, struct tplg_elem *elem) snd_config_iterator_t i, next; snd_config_t *n; struct snd_soc_tplg_ctl_tlv *tplg_tlv; + struct snd_soc_tplg_tlv_dbscale *scale; const char *id = NULL, *value = NULL; - int *data; tplg_dbg(" scale: %s\n", elem->id); tplg_tlv = calloc(1, sizeof(*tplg_tlv)); if (!tplg_tlv) return -ENOMEM; - data = (int*)(tplg_tlv->data); elem->tlv = tplg_tlv; - tplg_tlv->numid = SNDRV_CTL_TLVT_DB_SCALE; - tplg_tlv->count = 8; - tplg_tlv->size = sizeof(*tplg_tlv); + tplg_tlv->size = sizeof(struct snd_soc_tplg_ctl_tlv); + tplg_tlv->type = SNDRV_CTL_TLVT_DB_SCALE; + scale = &tplg_tlv->scale; snd_config_for_each(i, next, cfg) { @@ -242,11 +238,11 @@ static int tplg_parse_tlv_dbscale(snd_config_t *cfg, struct tplg_elem *elem) /* get TLV data */ if (strcmp(id, "min") == 0) - data[0] = atoi(value); + scale->min = atoi(value); else if (strcmp(id, "step") == 0) - data[1] = atoi(value); + scale->step = atoi(value); else if (strcmp(id, "mute") == 0) - data[2] = atoi(value); + scale->mute = atoi(value); else SNDERR("error: unknown key %s\n", id); } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: topology: Use kcalloc instead of kzalloc for array allocation
by Axel Lin 05 Aug '15

05 Aug '15

Also remove unnecessary memset. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/soc-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..a5eb48e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1250,7 +1250,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( struct snd_kcontrol_new *kc; int i, err; - kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + kc = kcalloc(count, sizeof(*kc), GFP_KERNEL); if (!kc) return NULL; @@ -1273,7 +1273,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - memset(kc, 0, sizeof(*kc)); kc[i].name = be->hdr.name; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; -- 2.1.0

1 0

[alsa-devel] [PATCH] ASoC: Add function "rl6231_get_pre_div" to correct the dmic clock calculation
by Oder Chiou 05 Aug '15

05 Aug '15

Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rl6231.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rl6231.h | 1 + sound/soc/codecs/rt5640.c | 7 ++++--- sound/soc/codecs/rt5645.c | 7 ++++--- sound/soc/codecs/rt5651.c | 7 ++++--- sound/soc/codecs/rt5670.c | 7 ++++--- sound/soc/codecs/rt5677.c | 5 ++++- 7 files changed, 68 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 96f3e90..57e51c1 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,10 +11,57 @@ */ #include <linux/module.h> +#include <linux/regmap.h> #include "rl6231.h" /** + * rl6231_get_pre_div - Return the value of pre divider. + * + * @map: map for setting. + * @reg: register. + * @sft: shift. + * + * Return the value of pre divider from given register value. + * Return negative error code for unexpected register value. + */ +int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft) +{ + int pd, val; + + regmap_read(map, reg, &val); + + val = (val >> sft) & 0x7; + + switch (val) { + case 0: + case 1: + case 2: + case 3: + pd = val + 1; + break; + case 4: + pd = 6; + break; + case 5: + pd = 8; + break; + case 6: + pd = 12; + break; + case 7: + pd = 16; + break; + default: + pd = -EINVAL; + break; + } + + return pd; +} +EXPORT_SYMBOL_GPL(rl6231_get_pre_div); + +/** * rl6231_calc_dmic_clk - Calculate the parameter of dmic. * * @rate: base clock rate. diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h index 0f7b057..4c77b44 100644 --- a/sound/soc/codecs/rl6231.h +++ b/sound/soc/codecs/rl6231.h @@ -30,5 +30,6 @@ int rl6231_calc_dmic_clk(int rate); int rl6231_pll_calc(const unsigned int freq_in, const unsigned int freq_out, struct rl6231_pll_code *pll_code); int rl6231_get_clk_info(int sclk, int rate); +int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft); #endif /* __RL6231_H__ */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 119eca7..c557192 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -459,10 +459,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5640->sysclk); + int idx, rate; + rate = rt5640->sysclk / rl6231_get_pre_div(rt5640->regmap, + RT5640_ADDA_CLK1, RT5640_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d54b34..4b01b63 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -543,10 +543,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5645->sysclk); + int idx, rate; + rate = rt5645->sysclk / rl6231_get_pre_div(rt5645->regmap, + RT5645_ADDA_CLK1, RT5645_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 96295cc..8f7159b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -378,10 +378,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5651->sysclk); + int idx, rate; + rate = rt5651->sysclk / rl6231_get_pre_div(rt5651->regmap, + RT5651_ADDA_CLK1, RT5651_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index fd6ffd2..177748a 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -683,10 +683,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5670->sysclk); + int idx, rate; + rate = rt5670->sysclk / rl6231_get_pre_div(rt5670->regmap, + RT5670_ADDA_CLK1, RT5670_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..3316700 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -916,8 +916,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int idx = rl6231_calc_dmic_clk(rt5677->lrck[RT5677_AIF1] << 8); + int idx, rate; + rate = rt5677->sysclk / rl6231_get_pre_div(rt5677->regmap, + RT5677_CLK_TREE_CTRL1, RT5677_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] Segmentation Fault in snd_pcm_rate_hw_free()
by Valentin Corfu 05 Aug '15

05 Aug '15

Hello ALSA developers, I observed one segmentation fault in snd_pcm_rate_hw_free() function, with the following BT: (gdb) up #1 0xb7554cc1 in raise (sig=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:64 64 return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig); (gdb) #2 0xb75580ee in abort () at abort.c:92 92 raise (SIGABRT); (gdb) #3 0xb758a7dd in __libc_message (do_abort=2, fmt=0xb766053c "*** glibc detected *** %s: %s: 0x%s ***\n") at ../sysdeps/unix/sysv/linux/libc_fatal.c:189 189 abort (); (gdb) #4 0xb7594a71 in malloc_printerr (action=<value optimized out>, str=<value optimized out>, ptr=0x969ae98) at malloc.c:6283 6283 __libc_message (action & 2, (gdb) #5 0xb759636b in _int_free (av=<value optimized out>, p=0x969ae90) at malloc.c:4795 4795 malloc_printerr (check_action, errstr, chunk2mem(p)); (gdb) #6 0xb75994bd in __libc_free (mem=0x969ae98) at malloc.c:3738 3738 _int_free(ar_ptr, p); (gdb) #7 0xb76f3a81 in snd_pcm_rate_hw_free (pcm=0x9685d78) at pcm_rate.c:341 341 free(rate->pareas[0].addr); (gdb) #8 0xb76d045b in snd_pcm_hw_free (pcm=0x9685d78) at pcm.c:858 858 err = pcm->ops->hw_free(pcm->op_arg); (gdb) #9 0xb76f826e in snd_pcm_plug_hw_free (pcm=0x96856b0) at pcm_plug.c:1046 1046 int err = snd_pcm_hw_free(slave); (gdb) #10 0xb76d045b in snd_pcm_hw_free (pcm=0x96856b0) at pcm.c:858 858 err = pcm->ops->hw_free(pcm->op_arg); (gdb) #11 0x080492ad in main () Could you please give me some hints how to solve this issue? I can provide you more info or the test application, if needed. I can see the issue every time, and I also checked with latest version of alsa-lib but I got the same results. Thank you, Valentin Corfu

2 6

[alsa-devel] [PATCH] topology: Add support for writing manifest private data.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Allow manifest to contain private data and write this data to file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/builder.c | 37 ++++++++++++++++++++++++++++++++----- src/topology/tplg_local.h | 1 + 2 files changed, 33 insertions(+), 5 deletions(-) diff --git a/src/topology/builder.c b/src/topology/builder.c index 0066b22..a944866 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -226,9 +226,6 @@ static int write_block(snd_tplg_t *tplg, struct list_head *base, case OBJECT_TYPE_CC: return write_elem_block(tplg, base, size, SND_SOC_TPLG_TYPE_DAI_LINK, "cc"); - case OBJECT_TYPE_MANIFEST: - return write_data_block(tplg, size, SND_SOC_TPLG_TYPE_MANIFEST, - "manifest", &tplg->manifest); case OBJECT_TYPE_DATA: return write_elem_block(tplg, base, size, SND_SOC_TPLG_TYPE_PDATA, "data"); @@ -239,13 +236,43 @@ static int write_block(snd_tplg_t *tplg, struct list_head *base, return 0; } +/* write the manifest including its private data */ +static int write_manifest_data(snd_tplg_t *tplg) +{ + int ret; + + /* write the header for this block */ + ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, + SND_SOC_TPLG_ABI_VERSION, 0, + sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); + if (ret < 0) { + SNDERR("error: failed to write manifest block %d\n", ret); + return ret; + } + + verbose(tplg, "manifest : write %d bytes\n", sizeof(tplg->manifest)); + ret = write(tplg->out_fd, &tplg->manifest, sizeof(tplg->manifest)); + if (ret < 0) { + SNDERR("error: failed to write manifest %d\n", ret); + return ret; + } + + verbose(tplg, "manifest : write %d priv bytes\n", tplg->manifest.priv.size); + ret = write(tplg->out_fd, tplg->manifest_pdata, tplg->manifest.priv.size); + if (ret < 0) { + SNDERR("error: failed to write manifest priv data %d\n", ret); + return ret; + } + + return 0; +} + int tplg_write_data(snd_tplg_t *tplg) { int ret; /* write manifest */ - ret = write_data_block(tplg, sizeof(tplg->manifest), - OBJECT_TYPE_MANIFEST, "manifest", &tplg->manifest); + ret = write_manifest_data(tplg); if (ret < 0) { SNDERR("failed to write manifest %d\n", ret); return ret; diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 62788e4..ad38945 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -77,6 +77,7 @@ struct snd_tplg { /* manifest */ struct snd_soc_tplg_manifest manifest; + const void *manifest_pdata; /* copied by builder at file write */ /* list of each element type */ struct list_head tlv_list; -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: Add element ID so we can look up references by name.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Jin Yao <yao.jin(a)linux.intel.com> Add support to lookup elements by name. This is in preparation for adding some new API calls that will allow building topology data using a C API. This will allow applications to build thier own topology data directly. Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- src/topology/ctl.c | 9 ++++----- src/topology/dapm.c | 2 +- src/topology/data.c | 2 +- src/topology/elem.c | 15 +++++++++++---- src/topology/pcm.c | 10 +++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 7 files changed, 24 insertions(+), 18 deletions(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 9c1333c..aa06ff6 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -264,7 +264,7 @@ int tplg_parse_tlv(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_TLV); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_TLV); if (!elem) return -ENOMEM; @@ -298,7 +298,7 @@ int tplg_parse_control_bytes(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_BYTES); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_BYTES); if (!elem) return -ENOMEM; @@ -403,11 +403,10 @@ int tplg_parse_control_enum(snd_tplg_t *tplg, snd_config_t *cfg, const char *id, *val = NULL; int err, j; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_ENUM); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_ENUM); if (!elem) return -ENOMEM; - /* init new mixer */ ec = elem->enum_ctrl; elem_copy_text(ec->hdr.name, elem->id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); ec->hdr.type = SND_SOC_TPLG_TYPE_ENUM; @@ -501,7 +500,7 @@ int tplg_parse_control_mixer(snd_tplg_t *tplg, const char *id, *val = NULL; int err, j; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_MIXER); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_MIXER); if (!elem) return -ENOMEM; diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 1da82ad..7e26ea0 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -420,7 +420,7 @@ int tplg_parse_dapm_widget(snd_tplg_t *tplg, const char *id, *val = NULL; int widget_type, err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_DAPM_WIDGET); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_DAPM_WIDGET); if (!elem) return -ENOMEM; diff --git a/src/topology/data.c b/src/topology/data.c index 13e1e2b..c768bc5 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -268,7 +268,7 @@ int tplg_parse_data(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_DATA); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_DATA); if (!elem) return -ENOMEM; diff --git a/src/topology/elem.c b/src/topology/elem.c index d7a1fd7..7fee653 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -103,20 +103,27 @@ struct tplg_elem *tplg_elem_lookup(struct list_head *base, const char* id, /* create a new common element and object */ struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, - snd_config_t *cfg, enum object_type type) + snd_config_t *cfg, const char *name, enum object_type type) { struct tplg_elem *elem; const char *id; int obj_size = 0; void *obj; + if (!cfg && !name) + return NULL; + elem = tplg_elem_new(); if (!elem) return NULL; - snd_config_get_id(cfg, &id); - strncpy(elem->id, id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - elem->id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1] = 0; + /* do we get name from cfg */ + if (cfg) { + snd_config_get_id(cfg, &id); + elem_copy_text(elem->id, id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + elem->id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1] = 0; + } else if (name != NULL) + elem_copy_text(elem->id, name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); switch (type) { case OBJECT_TYPE_DATA: diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 8f23a6f..deae47b 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -228,7 +228,7 @@ int tplg_parse_pcm_config(snd_tplg_t *tplg, const char *id; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_STREAM_CONFIG); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_STREAM_CONFIG); if (!elem) return -ENOMEM; @@ -294,7 +294,7 @@ int tplg_parse_pcm_caps(snd_tplg_t *tplg, char *s; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_STREAM_CAPS); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_STREAM_CAPS); if (!elem) return -ENOMEM; @@ -461,7 +461,7 @@ int tplg_parse_pcm(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_PCM); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_PCM); if (!elem) return -ENOMEM; @@ -524,7 +524,7 @@ int tplg_parse_be(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_BE); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_BE); if (!elem) return -ENOMEM; @@ -587,7 +587,7 @@ int tplg_parse_cc(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_CC); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_CC); if (!elem) return -ENOMEM; diff --git a/src/topology/text.c b/src/topology/text.c index ebb6e38..7128056 100644 --- a/src/topology/text.c +++ b/src/topology/text.c @@ -64,7 +64,7 @@ int tplg_parse_text(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_TEXT); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_TEXT); if (!elem) return -ENOMEM; diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 688c78f..62788e4 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -216,7 +216,7 @@ struct tplg_elem *tplg_elem_lookup(struct list_head *base, const char* id, unsigned int type); struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, - snd_config_t *cfg, enum object_type type); + snd_config_t *cfg, const char *name, enum object_type type); static inline void elem_copy_text(char *dest, const char *src, int len) { -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: fix element object type is switch()
by Liam Girdwood 05 Aug '15

05 Aug '15

Use the correct type for this object. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/elem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/elem.c b/src/topology/elem.c index 32ba2c1..d7a1fd7 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -137,7 +137,7 @@ struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, list_add_tail(&elem->list, &tplg->enum_list); obj_size = sizeof(struct snd_soc_tplg_enum_control); break; - case SND_SOC_TPLG_TYPE_MIXER: + case OBJECT_TYPE_MIXER: list_add_tail(&elem->list, &tplg->mixer_list); obj_size = sizeof(struct snd_soc_tplg_mixer_control); break; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix Dell laptop for internal mic/headset mic
by woodrow.shen＠canonical.com 05 Aug '15

05 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> The new Dell laptop with ALC3266 can't use internal microphone to record. Also, the 3 ring headset mic doesn't work on it. For src of patch_realtek.c, only the pin cfg of Nid 0x17 is different, so the Nid 0x17 is added into pin quirk, and others are defined to macro ALC298_STANDARD_PINS in common use. Codec: Realtek ALC3266 Vendor Id: 0x10ec0298 Subsystem Id: 0x102806e5 BugLink: https://bugs.launchpad.net/bugs/1481575 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b20f104..294ba67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5401,6 +5401,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x14, 0x90170110}, \ {0x15, 0x0221401f} +#define ALC298_STANDARD_PINS \ + {0x12, 0x90a60130}, \ + {0x21, 0x03211020} + static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, @@ -5570,9 +5574,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC292_STANDARD_PINS, {0x13, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60130}, - {0x17, 0x90170140}, - {0x21, 0x03211020}), + ALC298_STANDARD_PINS, + {0x17, 0x90170140}), + SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC298_STANDARD_PINS, + {0x17, 0x90170110}), {} }; -- 2.1.4

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Re: [alsa-devel] When select 2.1 profile in pulseaudio, how do pulseaudio select notebook 's internal subwoofer or 5.1 external speaker ?
by Raymond Yau 05 Aug '15

05 Aug '15

> > > > Do pulseaudio 2.1 profile select the internal subwoofer or your 5.1 > > external speaker ? > > My external speaker (i.e not subwoofer but center). It always was so. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… +/* default channel maps for 2.1 speakers; + * since HD-audio supports only stereo, odd number channels are omitted + */ +const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_LFE } }, + { } +}; The default 2.1 channel map for internal subwoofer does not has 5.1 channel map But your notebook have three jacks at ext front which support 5.1 In your case,there is no default 2.1 channel map when the current model has three speaker pins after remove the not exist internal surround speakers, [ALC882_FIXUP_ACER_ASPIRE_4930G] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x99130111 }, /* CLFE speaker */ - { 0x17, 0x99130112 }, /* surround speaker */ - { } - }, you need to use the left channel version of 2.1 channel and 5.1 channel map +static const struct snd_pcm_chmap_elem acer_pcm_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_NA } }, /* LFE only on left */ + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e… <http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e…> conf: Allow 2.1 surround to use different number of channels This way, cards that support LFE on four channels (e g laptop with internal subwoofer) can do that, and other cards on a six channel setup can use that as well. which channel map is used by alsa lib when you open surround21 pcm device since your notebook will have both channel maps ?

1 1

[alsa-devel] [RFC][PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk
by Takashi Sakamoto 05 Aug '15

05 Aug '15

Some users have reported current ALSA Fireworks driver stops packet streaming due to detecting discontinuity. [alsa-devel] ALSA snd-fireworks Echo AudioFire 2 http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [alsa-devel] AudioFire4 quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095697.html As long as I investigate, this issue depends on firmware version installed in these devices. While, it's not clear that which version of firmwares for which model has this quirk. At least, firmware version 5.7.0, 5.7.3 and 5.8.0 for AudioFire2/AudioFire4/AudioFirePre8 have this quirk. This patchset applies a solution for this issue, with my assumption that 'the same firmware binary has the same quirk'. These models, AudioFire8 (since Jul 2009) and Gibson Robot Interface Pack (RIP) series uses the same firmware binary in driver package. Takashi Sakamoto (2): Revert "ALSA: fireworks: add support for AudioFire2 quirk" ALSA: fireworks/firewire-lib: add support for recent firmware quirk sound/firewire/amdtp.c | 5 +++-- sound/firewire/amdtp.h | 2 ++ sound/firewire/fireworks/fireworks.c | 10 ++++++++-- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_stream.c | 12 +++++++++--- 5 files changed, 23 insertions(+), 8 deletions(-) -- 2.1.4

2 5

Re: [alsa-devel] ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()
by Dan Carpenter 04 Aug '15

04 Aug '15

Hello Lars-Peter Clausen, The patch 1ce43acff0c0: "ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()" from Jul 26, 2015, leads to the following static checker warning: sound/soc/soc-dapm.c:1199 snd_soc_dapm_dai_get_connected_widgets() warn: inconsistent returns 'mutex:&card->dapm_mutex'. Locked on: line 1194 Unlocked on: line 1199 sound/soc/soc-dapm.c 1164 int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, 1165 struct snd_soc_dapm_widget_list **list) 1166 { 1167 struct snd_soc_card *card = dai->component->card; 1168 struct snd_soc_dapm_widget *w; 1169 LIST_HEAD(widgets); 1170 int paths; 1171 int ret; 1172 1173 mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); 1174 1175 /* 1176 * For is_connected_{output,input}_ep fully discover the graph we need 1177 * to reset the cached number of inputs and outputs. 1178 */ 1179 list_for_each_entry(w, &card->widgets, list) { 1180 w->inputs = -1; 1181 w->outputs = -1; 1182 } 1183 1184 if (stream == SNDRV_PCM_STREAM_PLAYBACK) 1185 paths = is_connected_output_ep(dai->playback_widget, &widgets); 1186 else 1187 paths = is_connected_input_ep(dai->capture_widget, &widgets); 1188 1189 /* Drop starting point */ 1190 list_del(widgets.next); 1191 1192 ret = dapm_widget_list_create(list, &widgets); 1193 if (ret) 1194 return ret; ^^^^^^^^^^^ Holding card->dapm_mutex. 1195 1196 trace_snd_soc_dapm_connected(paths, stream); 1197 mutex_unlock(&card->dapm_mutex); 1198 1199 return paths; 1200 } regards, dan carpenter

2 1