June 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v1 0/4] ASoC: Codecs: Add TDA7802 codec
by Thomas Preston 14 Jun '19

14 Jun '19

This patch series adds a driver for the ST TDA7802 amplifier. Thanks, Thomas Cc: Patrick Glaser <pglaser(a)tesla.com> Cc: Rob Duncan <rduncan(a)tesla.com> Cc: Nate Case <ncase(a)tesla.com> Thomas Preston (4): dt-bindings: ASoC: Add TDA7802 amplifier ASoC: Add codec driver for ST TDA7802 ASoC: tda7802: Add enable device attribute ASoC: tda7802: Add speaker-test sysfs .../devicetree/bindings/sound/tda7802.txt | 26 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7802.c | 815 +++++++++++++++++++++ 4 files changed, 849 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7802.txt create mode 100644 sound/soc/codecs/tda7802.c -- 2.11.0

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[alsa-devel] [PATCH 0/3] Add mclk0 clock source for SAI
by Daniel Baluta 14 Jun '19

14 Jun '19

This patch series brings together patches [1] and [2] which introduce mclk0 clock source via DT. mclk0..3 are the four clock sources options of SAI's clock MUX. mclk0 option selects: - Bus Clock on i.MX8 - MCLK1 on i.MX6/7 Finally we also update the DT binding information for SAI clocks. In [1] and [2] Nicolin had a very good point on the fact that mclk0 might not be needed in the DT. Anyhow, there are two reasons for which I think mlck0 should be added to DT: 1) SAI clock source select MUX is really part of the hardware 2) flexibility! We let DT tell us which is the option for MUX option 0. [1] lkml.org/lkml/2019/4/20/141 [2] lkml.org/lkml/2019/4/20/56 Daniel Baluta (2): dt-bindings: sound: Clarify the usage of clocks in SAI ASoC: fsl_sai: Read SAI clock source 0 from DT Shengjiu Wang (1): ARM: dts: imx: Add mclk0 clock for SAI Documentation/devicetree/bindings/sound/fsl-sai.txt | 5 +++-- arch/arm/boot/dts/imx6sx.dtsi | 6 ++++-- arch/arm/boot/dts/imx6ul.dtsi | 9 ++++++--- arch/arm/boot/dts/imx7s.dtsi | 9 ++++++--- sound/soc/fsl/fsl_sai.c | 3 +-- 5 files changed, 20 insertions(+), 12 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v6 0/4] Fix jack detection for Chromebook Pixel
by Fletcher Woodruff 14 Jun '19

14 Jun '19

** resending due to corrupted patch ** Headphone/mic jack detection doesn't work on the Chromebook Pixel 2015. This patch changes the irq implementation to support polarity flipping and fixes the configuration code so that correct GPIO pins are read from ACPI. With this series, plugging and unplugging the headphone jack switches between headphones and speakers automatically, and headset microphones are also detected. v6: - Move refactoring into its own patch - Reorder patches so that DT property names patch is first - Clarify commit message for patch which implements irq handler - Remove unused work struct - Make IRQ function return IRQ_HANDLED only if IRQs actually fire v5: - Fix void* parameter to devm_request_threaded_irq v4: - Fix incorrect void* cast in rt5677_irq() v3: - Update commit message for patch 1/3 to clarify why we implement our own irq_chip. v2: - Split IRQ change into two patches: adding and fixing potential race - Change config reading code to try both DT and ACPI style names Ben Zhang (2): ASoC: rt5677: clear interrupts by polarity flip ASoC: rt5677: handle concurrent interrupts Fletcher Woodruff (2): ASoC: rt5677: fall back to DT prop names on error ASoC: rt5677: move jack-detect init to i2c probe sound/soc/codecs/rt5677.c | 319 ++++++++++++++++++++++++++------------ sound/soc/codecs/rt5677.h | 13 +- 2 files changed, 236 insertions(+), 96 deletions(-) -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH v2 0/3] ASoC: SOF: suspend/resume debug tools
by Pierre-Louis Bossart 14 Jun '19

14 Jun '19

These patches were submitted in an earlier series but not merged and then conflicted with the IPC flood test. Resending, no new functionality. These helpers are really helpful for e.g. SoundWire or HDaudio bringup. Pierre-Louis Bossart (3): ASoC: SOF: trace: move to opt-in with Kconfig and module parameter ASoC: SOF: acpi: add module param to disable pm_runtime ASoC: SOF: pci: add module param to disable pm_runtime sound/soc/sof/Kconfig | 8 ++++++++ sound/soc/sof/core.c | 26 ++++++++++++++++++++------ sound/soc/sof/sof-acpi-dev.c | 12 +++++++++++- sound/soc/sof/sof-pci-dev.c | 12 +++++++++++- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/trace.c | 17 ++++++++++++++++- 6 files changed, 67 insertions(+), 9 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] Revert "ALSA: hda/realtek - Improve the headset mic for Acer Aspire laptops"
by Hui Wang 14 Jun '19

14 Jun '19

This reverts commit 9cb40eb184c4220d244a532bd940c6345ad9dbd9. This patch introduces noise and headphone playback issue after rebooting or suspending/resuming. Let us revert it. https://bugzilla.kernel.org/show_bug.cgi?id=203831 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1afb268f3da0..179e4be1f747 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6268,15 +6268,13 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC255_FIXUP_ACER_MIC_NO_PRESENCE] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* Enable the Mic */ - { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, - {} + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } }, .chained = true, - .chain_id = ALC269_FIXUP_LIFEBOOK_EXTMIC + .chain_id = ALC255_FIXUP_HEADSET_MODE }, [ALC255_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -7320,10 +7318,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x18, 0x02a11030}, {0x19, 0x0181303F}, {0x21, 0x0221102f}), - SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x14, 0x90170120}, - {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE, {0x12, 0x90a601c0}, {0x14, 0x90171120}, -- 2.17.1

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[alsa-devel] PortAudio/Alsa devices question on linux
by Mark Moreno 14 Jun '19

14 Jun '19

I built portaudio on rh75 in its normal 64 bit mode and when I run its test program pa_devs which lists the audio devices available, my usb sound card device is listed and my application that uses portaudio works fine. I built portaudio on rh75 in 32 bit form and when I run its test program pa_devs to list the devices, it doesn't list my usb sound card and so my application doesn't work. Why would a usb audio sound card be seen by a 64bit version of portaudio and not by a 32bit version of portaudio? I assume it's using alsa under the hood so not sure why there would be any difference. Am I missing something on my RH75 linux install or ??? thanksMark Moreno

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[alsa-devel] [PATCH] ASoC: nau8825: fix fake interruption when booting
by John Hsu 14 Jun '19

14 Jun '19

There is no pull-up resistor at IRQ line where it connects from the codec to SoC. When booting, the signal of IRQ pin will keep low which makes the SoC invoke the ISR repeatedly because the IRQ is registered trigger low. It will not stop until the codec sets up the interruption and pulls the signal high. In the patch, nau8825 will internally pull the signal to high at booting in case the fake interrupts happen. The patch help to fix the issue as follows: Google issue 133904101: Cave: excessive number of interrupts before Nau8825 codec is initialized Reported Issue Chrome OS Version : ChromeOS R76-12217.0.0 Type of hardware : What steps will reproduce the problem? Boot device What is the expected output? A lot of interrupts without any events What do you see instead? The fake interrupts are gone How frequently does this problem reproduce? Always What is the impact to the user, and is there a workaround? If so, what is it? BUG=b:133904101 TEST=Boot device, and then check the interrupts are real. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 4 ++++ sound/soc/codecs/nau8825.h | 2 ++ 2 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 47e65cf99879..83ec841f7865 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1881,6 +1881,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_JACK_EJECT_DEBOUNCE_MASK, nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + /* Pull up IRQ pin */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN, + NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN); /* Mask unneeded IRQs: 1 - disable, 0 - enable */ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index f6074c618569..3f41897ed3f6 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -171,6 +171,8 @@ #define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ /* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_PIN_PULLUP (1 << 14) +#define NAU8825_IRQ_PIN_PULL_EN (1 << 13) #define NAU8825_IRQ_OUTPUT_EN (1 << 11) #define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) #define NAU8825_IRQ_RMS_EN (1 << 8) -- 2.21.0

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[alsa-devel] [PATCH] ucm: Dell WD15 set lineout controls
by Curtis Malainey 14 Jun '19

14 Jun '19

The lineout playback switch should be enabled and disabled when enabling and disabling the device. This commit needs to be tested by dell. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Crag Wang <crag.wang(a)dell.com> --- src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf index 1b6bd212..cab13c5d 100644 --- a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -1,3 +1,11 @@ +SectionVerb { + EnableSequence [ + cdev "hw:Dock" + ] + DisableSequence [ + ] +} + SectionDevice."Headphone" { Comment "Headphone" @@ -14,6 +22,14 @@ SectionDevice."LineOut" { PlaybackChannels "2" PlaybackPCM "hw:Dock,1" } + EnableSequence [ + cdev "hw:Dock" + cset "name='Line Playback Switch' on" + ] + DisableSequence [ + cdev "hw:Dock" + cset "name='Line Playback Switch' off" + ] } SectionDevice."Mic" { -- 2.22.0.rc2.383.gf4fbbf30c2-goog

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Re: [alsa-devel] (no subject)
by Ralf Beck 13 Jun '19

13 Jun '19

Jaroslav Kysela wrote: >I just don't think that the mmap transfer mode is the culprit of the problems. >The problem is that the PCI cards off-loads the DMA transfers completely >without the extra framing required for those serial hardware interfaces which >are handled in the audio driver and the other kernel stacks (USB etc.). The source of all evil is the ring buffer. It prevents all devices that require packet headers (Firewire/network based solutions) or that use a variable amount of frames per packet (all asynchronous devices, especially asynchronous USB ISO endpoints) from mapping their data directly into userspas memory without ever touching the audio data in it. While this doesn't hurt performance to much with devices that offer only a few channels, it has a massive effect on network based devices with possibly hundreds of channels of which only a small number is used (e.g. record enabled) at a time. The ringbuffer should be replaced by a buffer that holds the packets and a descriptor for how to find the data in it. Btw, alsa uses something similar already for the mmap handling. The descriptor should consist of x periods, containing y chunks with a max size, containing z channels, each of the channel described by a start address and step value within that chunk. Examples: PCI device; 2 periods, 1 chunks per period, max size 64 frames per chunk, 2 channels per chunk USB device: 2 periods, 8 chunks (microframes) max size 8 frames (if at 44.1/48kHz), 2 channels per chunk, Same for Firewire, AVB (each chunk possibly containing several AVB streams, i.e. ethernet packets), etc. Period elapsed => the number of chunks forming a period have been received snd_pcm_avail(_update) => return the sum of actual frames of the period snd_pcm_mmap_commit => return the descriptor for one chunk, the number of frames of the chunk and a channel map for the chunk. The reason for the latter is that on ethernet based devices (AVB) there is bo guarantee that streams of different endpoints are received in the same order (but in the same interval window). Alsa clients should then loop until they have processed all frames thar have been reported by snd_pcm_avail (instead of a period size number of frames). Ah, and AM824 raw audio format, used by firewire and AVB, should by added to the list of audio formats, so conversion can be done in userspace. Ralf

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[alsa-devel] [PATCH 0/4] ASoC: meson: axg-tdm: i2s format fixups
by Jerome Brunet 13 Jun '19

13 Jun '19

The patchset fixes a few mistakes regarding the format on the i2s bus on the Amlogic axg tdm drivers. Jerome Brunet (4): ASoC: meson: axg-tdmin: right_j is not supported ASoC: meson: axg-tdmout: right_j is not supported ASoC: meson: axg-tdm: fix sample clock inversion ASoC: meson: axg-tdm: consistently use SND_SOC_DAIFMT defines sound/soc/meson/axg-tdm-interface.c | 4 ++-- sound/soc/meson/axg-tdm.h | 2 +- sound/soc/meson/axg-tdmin.c | 1 - sound/soc/meson/axg-tdmout.c | 1 - 4 files changed, 3 insertions(+), 5 deletions(-) -- 2.20.1

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