January 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/5] ASOC: bcm2835: move bcm2835-i2s to use clock framework
by kernel＠martin.sperl.org 11 Jan '16

11 Jan '16

From: Martin Sperl <kernel(a)martin.sperl.org> This patchset enables the bcm2835-i2s driver to use the clock framework which was introduced with commit 94cb7f76caa0b337 ("Switch to using the new clock driver support"). This commit resulted in the fact that the bcm2835-i2s driver was no longer working due to some register addresses used by 2 drivers (clk-bcm2835 and bcm2835-i2s). To make it all possible this also required the introduction of the PCM clock into the clk-bcm2835 driver. This patchset relies on the patch by Remi Pommarel <repk(a)triplefau.lt> that introduces the ability to set parent clocks ("clk: bcm2835: Support for clock parent selection"), which is (as far as I understood) in clk-next and slated for 4.5. Note that there is one regression: the clk-bcm2835 does not yet support the mash functionality which the SOC-Hw supports, this may result in slightly more "audiable noise" than the original driver. But as this is more about making the driver functional again, this is - I believe - a drawback we can accept for now. Martin Sperl (5): ASoC: bcm2835: cleanup includes by ordering them alphabetically clk: bcm2835: enable management of PCM clock ASoC: bcm2835: move to use the clock framework ARM: bcm2835: I2S: use new register-range and clock framework dt-bindings: bsm2835: fix bindings documentation to use new clock framework .../devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 7 +- arch/arm/boot/dts/bcm2835.dtsi | 5 +- drivers/clk/bcm/clk-bcm2835.c | 15 + include/dt-bindings/clock/bcm2835.h | 3 +- sound/soc/bcm/bcm2835-i2s.c | 293 +++++--------------- 5 files changed, 91 insertions(+), 232 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH -next] ASoC: AMD: Add missing include file
by Guenter Roeck 11 Jan '16

11 Jan '16

arm:allmodconfig, s390:allmodconfig, sparc64:allmodconfig, and probably other builds fail with sound/soc/amd/acp-pcm-dma.c:83:2: error: implicit declaration of function ‘readl’ sound/soc/amd/acp-pcm-dma.c:88:2: error: implicit declaration of function ‘writel’ Include linux/io.h explicitly to fix the problem. Fixes: 7c31335a03b6a ("ASoC: AMD: add AMD ASoC ACP 2.x DMA driver") Cc: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/amd/acp-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cc8b841b69b6..3191e0a7d273 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/io.h> #include <linux/sizes.h> #include <linux/pm_runtime.h> -- 2.1.4

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
by Arnd Bergmann 11 Jan '16

11 Jan '16

A recent rework removed the only user of the hdac_hdmi_query_pin_connlist function, so we now get a warning when building the hdac_hdmi driver: hdac_hdmi.c:313:12: warning: 'hdac_hdmi_query_pin_connlist' defined but not used [-Wunused-function] This removes the function, which makes the file build cleanly again. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 15b914476bf2 ("ASoC: hdac_hdmi: Use list to add pins and converters") --- I saw this warning the first time on today's linux-next with ARM randconfig builds diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f5df7232405b..5a1ec0f7a1a6 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -310,28 +310,6 @@ hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt) return err; } -static int hdac_hdmi_query_pin_connlist(struct hdac_ext_device *hdac, - struct hdac_hdmi_pin *pin) -{ - if (!(get_wcaps(&hdac->hdac, pin->nid) & AC_WCAP_CONN_LIST)) { - dev_warn(&hdac->hdac.dev, - "HDMI: pin %d wcaps %#x does not support connection list\n", - pin->nid, get_wcaps(&hdac->hdac, pin->nid)); - return -EINVAL; - } - - pin->num_mux_nids = snd_hdac_get_connections(&hdac->hdac, pin->nid, - pin->mux_nids, HDA_MAX_CONNECTIONS); - if (pin->num_mux_nids == 0) - dev_warn(&hdac->hdac.dev, "No connections found for pin: %d\n", - pin->nid); - - dev_dbg(&hdac->hdac.dev, "num_mux_nids %d for pin: %d\n", - pin->num_mux_nids, pin->nid); - - return pin->num_mux_nids; -} - static void hdac_hdmi_fill_widget_info(struct snd_soc_dapm_widget *w, enum snd_soc_dapm_type id, const char *wname, const char *stream)

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[alsa-devel] [PATCH] ASoC: AMD: add missing include
by Arnd Bergmann 11 Jan '16

11 Jan '16

The newly added AMD acp driver is missing an include for linux/io.h and fails to build on ARM: sound/soc/amd/acp-pcm-dma.c: In function 'acp_reg_read': sound/soc/amd/acp-pcm-dma.c:83:9: error: implicit declaration of function 'readl' [-Werror=implicit-function-declaration] This adds the missing line. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This build error also appeared today on linux-next with ARM randconfig diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cc8b841b69b6..fa0311e4faed 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/sizes.h> #include <linux/pm_runtime.h> +#include <linux/io.h> #include <sound/soc.h>

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[alsa-devel] [PATCH] ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
by David Henningsson 11 Jan '16

11 Jan '16

Inform userspace that one channel of the internal mic has reversed polarity, so it does not attempt to add both channels together and end up with silence. Cc: stable(a)vger.kernel.org Reported-by: Andrzej Mendel <andrzej.mendel(a)gmail.com> Alsa-info: http://www.alsa-project.org/db/?f=3088f82a0cf977855f92af9db8ad406c04f71efa BugLink: https://bugs.launchpad.net/bugs/1529624 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a89d82..2fdda51 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4666,6 +4666,7 @@ enum { ALC290_FIXUP_SUBWOOFER, ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, + ALC269_FIXUP_DMIC_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, @@ -5103,6 +5104,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, }, + [ALC269_FIXUP_DMIC_THINKPAD_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5457,6 +5464,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), + SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

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[alsa-devel] [PATCH] usb: Add native DSD support for Oppo HA-1
by Jurgen Kramer 11 Jan '16

11 Jan '16

This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset but they use their own vendor ID. 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 b6c0c8e..23ea6d8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1269,6 +1269,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ + case USB_ID(0x22d8, 0x0416): /* OPPO HA-1*/ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.5.0

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[alsa-devel] [PATCH] hda-emu: Add keycode* to input_dev
by David Henningsson 11 Jan '16

11 Jan '16

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/linux/input.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/linux/input.h b/include/linux/input.h index 213c12b..c710fe4 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -26,6 +26,11 @@ struct input_dev { unsigned long keybit[0x300 / (sizeof(long) * 8)]; /* ... */ + unsigned int keycodemax; + unsigned int keycodesize; + void *keycode; + /* ... */ + int (*open)(struct input_dev *dev); void (*close)(struct input_dev *dev); int (*flush)(struct input_dev *dev, struct file *file); -- 1.9.1

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[alsa-devel] [PATCH v3 0/5] Add support for Cirrus Logic CS47L24 and WM1831 codecs
by Richard Fitzgerald 11 Jan '16

11 Jan '16

---- Changes since v2 MFD: - Make binding doc a separate patch, add updates to supply bindings - Fix review comments ASoC:: - Fix sort order in KConfig and Makefile MAINTAINERS: - Add patch to update MAINTAINERS ---- mfd patch based on Lee Jones's for-next gpio patch based on Linus Walleij's for-next ASoC patch based on Mark Brown's for-next Lee/Mark - can one of you take all three patches (with maintainer acks) so that the patch set is kept together? Richard Fitzgerald (5): mfd: arizona: Support Cirrus Logic CS47L24 and WM1831 mfd: arizona: Update DT bindings to add CS47L24 and WM1831 gpio: arizona: Support Cirrus Logic CS47L24 and WM1831 ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs MAINTAINERS: Update Wolfson Micro section to include CS47L24 source Documentation/devicetree/bindings/mfd/arizona.txt | 9 +- MAINTAINERS | 2 + drivers/gpio/gpio-arizona.c | 4 + drivers/mfd/Kconfig | 18 +- drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 74 +- drivers/mfd/arizona-irq.c | 40 +- drivers/mfd/arizona-spi.c | 7 + drivers/mfd/arizona.h | 4 + drivers/mfd/cs47l24-tables.c | 1629 +++++++++++++++++++++ include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l24.c | 1148 +++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 15 files changed, 2948 insertions(+), 26 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.9.1

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[alsa-devel] [PATCH] add power management S3 solution
by John Hsu 11 Jan '16

11 Jan '16

This patch is to fix interrupt fails when resume back. In solution, we add IRQ re-initiation, and reconstruct suspend and resume with set_bias_level function. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 36 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 504c969..2e2e11a 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1092,6 +1092,36 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* 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. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + nau8825_restart_jack_detection(regmap); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1121,6 +1151,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + if (nau8825->irq) + nau8825_resume_setup(nau8825); } break; @@ -1330,24 +1362,22 @@ static int nau8825_i2c_remove(struct i2c_client *client) #ifdef CONFIG_PM_SLEEP static int nau8825_suspend(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - disable_irq(client->irq); + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); return 0; } static int nau8825_resume(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); + enable_irq(nau8825->irq); return 0; } -- 2.6.4

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[alsa-devel] wrong channel mapping in a52 plugin
by Tom Yan 10 Jan '16

10 Jan '16

I am using alsa-lib/alsa-utils/alsa-plugins 1.1.0 and ffmpeg 2.8.4 and in Arch Linux. So I ran `speaker-test -D wav -c 6 -l 1` with the following .asoundrc: pcm.file { type file file st.ac3 } pcm.a52 { type plug slave { pcm { type a52 slavepcm file } } } pcm.wav { type file slave { pcm a52 } file st.wav format wav } [tom@localhost ~]$ speaker-test -D wav -c 6 -l 1 speaker-test 1.1.0 Playback device is wav Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 3072 to 206157312 Period size range from 1536 to 1536 Using max buffer size 206157312 Periods = 4 was set period_size = 1536 was set buffer_size = 206157312 0 - Front Left 2 - Front Center 1 - Front Right 5 - Rear Right 4 - Rear Left 3 - LFE Time per period = 0.203625 Then, I encode "st.wav" manually with `ffmpeg -i st.wav ff.ac3`. This shows a comparison between the wav file written directly from the speaker-test directly (st.wav) and the ac3 file I encode with a manual ffmpeg command (ff.ac3): https://ptpb.pw/5XJR.png As you can see, everything is preserved but the LFE is made LFE. And this shows a comparision between the the ac3 file from ffmpeg and the one written from the output of a52 plugin: https://ptpb.pw/YaKg.png If you compare carefully, you can see that Front Center and Rear Left is swapped, LFE and Rear Right is also swapped. Here is a pair of ac3 samples that shows the problem clearly (a lady will be telling the channels one by one): Original: https://ptpb.pw/9XdC.ac3 ALSA a52 plugin re-encoded from mpv output: https://ptpb.pw/SQ_7.ac3

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