August 2015 - Alsa-devel - mailman.alsa-project.org

[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(+)

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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