November 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [GIT PULL] Additional firmware files for CA0132 HD-audio codec
by Connor McAdams 14 Dec '18

14 Dec '18

The following changes since commit c6b6265d718d118e28e1ce8f91769aa886b54c94: Merge tag 'iwlwifi-fw-2018-10-03' of git://git.kernel.org/pub/scm/linux/kernel/git/iwlwifi/linux-firmware (2018-10-08 09:23:53 -0400) are available in the git repository at: git@github.com:Conmanx360/linux-firmware.git for you to fetch changes up to cd8899323352e772c5989230f35b1c494e0ab196: linux-firmware: Add new firmware for Creative CA0132 HD-Audio Codec (2018-10-09 14:41:12 -0400) ---------------------------------------------------------------- These are additional firmware files for CA0132 based sound cards. Creative has given me permission through email to send these to the linux-firmware repository under the same license as the previous ca0132 firmware. I asked my contact if they would sign-off on it, but they said to go ahead and submit it and that if there was any issue to contact them. If you need to contact them, or if there's more info needed, let me know and I can get see what I can do. ---------------------------------------------------------------- Connor McAdams (1): linux-firmware: Add new firmware for Creative CA0132 HD-Audio Codec WHENCE | 2 ++ ctefx-desktop.bin | Bin 0 -> 655856 bytes ctefx-r3di.bin | Bin 0 -> 655816 bytes 3 files changed, 2 insertions(+) create mode 100644 ctefx-desktop.bin create mode 100644 ctefx-r3di.bin

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[alsa-devel] [PATCH v3 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 14 Dec '18

14 Dec '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v2 (//lkml.org/lkml/2018/9/26/461): - Removed some unnecessary empty line adds. - Added more info to snd_compr_set_runtime_buffer() commit message Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 372 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 438 insertions(+), 4 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: core: Invoke pcm_new() for all DAI-link
by Rohit kumar 13 Dec '18

13 Dec '18

Remove no_pcm check to invoke pcm_new() for backend dai-links too. This fixes crash in hdmi codec driver during hdmi_codec_startup() while accessing chmap_info struct. chmap_info struct memory is allocated in pcm_new() of hdmi codec driver which is not invoked in case of DPCM when hdmi codec driver is part of backend dai-link. Below is the crash stack: [ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 .. [ 61.666696] CM = 0, WnR = 1 [ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [ 61.740269] PC is at hdmi_codec_startup+0x124/0x164 [ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164 Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12..abdc460 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1467,7 +1467,7 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, for (i = 0; i < num_dais; ++i) { struct snd_soc_dai_driver *drv = dais[i]->driver; - if (!rtd->dai_link->no_pcm && drv->pcm_new) + if (drv->pcm_new) ret = drv->pcm_new(rtd, dais[i]); if (ret < 0) { dev_err(dais[i]->dev, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [RFC PATCH v8 0/4] Media Device Allocator API
by shuah＠kernel.org 09 Dec '18

09 Dec '18

From: Shuah Khan <shuah(a)kernel.org> This patch series has been on the ice for the last couple of years. Mauro asked me to restart the discussion on this and see if we can make progress. I rebased the series to Linux 4.19. It has been surprisingly well preserved with one small merge conflict resolved in 0004. Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Please review. I am sending this as RFC even though it has been tested several times prior to this non-event rebased v8. I ran sanity tests. Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 +++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 320 +++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 692 insertions(+), 28 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.0

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[alsa-devel] [PATCH 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 08 Dec '18

08 Dec '18

Hi, this is patch series I've hacked after useful conversation with Pavel. Basically this adds a new LED trigger audio-mute and audio-micmute, and convert the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3, so that it can be merged well to multiple trees. Once after getting the ACK's, I'll add tags and fixate for merges. This patch series don't include huawei-wmi stuff; so Huawei patches need rework. I already have some piece of changes for huawei-wmi, so please ping me if needed. I checked briefly on my Dell laptop, and a Thinkpad model. Wider tests are appreciated, of course. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 45 +++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 +++++++++--- drivers/platform/x86/thinkpad_acpi.c | 66 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

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Re: [alsa-devel] HDA headset button support
by Takashi Iwai 07 Dec '18

07 Dec '18

On Wed, 28 Nov 2018 13:03:58 +0100, Kailang wrote: > > Hi Takashi, > > >I'd rather like to extend snd_hda_jack_add_kctl() with the key > >support (maybe adding a new function). You can pass there the > >combination of SND_JACK_BTN_X with KEY_XXX in an array. > > Do you mean create new function like below? > snd_hda_jack_add_key_kctl() > > It couldn't use snd_hda_jack_add_kctl() directly. > Because it had report pin state and get type from NID. > > type = get_input_jack_type(codec, nid); > ..... > ..... > state = snd_hda_jack_detect(codec, nid); > snd_jack_report(jack->jack, state ? jack->type : 0); > > So, to create a new function in hda_jack.c was better. Well, this seems more deeper than I thought. Below is my quick attempt to add the infrastructure. This adds more flexibility but still easiness for other possible button implementations, hopefully. Could you check the series below? thanks, Takashi

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[alsa-devel] [PATCH v8 0/7] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 07 Dec '18

07 Dec '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v8 - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp Vasily Khoruzhick (7): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget ASoC: dt-bindings: make enable-gpio optional for simple amplifier ASoC: simple-amplifier: make enable-gpio optional arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 35 ++++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 53 ++++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 35 ++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/codecs/simple-amplifier.c | 7 ++- 6 files changed, 188 insertions(+), 1 deletion(-) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by Niklas Carlsson 06 Dec '18

06 Dec '18

DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

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[alsa-devel] [PATCH] MAINTAINERS: add ASoC maintainers for sound dt-bindings
by Clément Péron 06 Dec '18

06 Dec '18

Sound dt-bindings are applied by ASoC maintainers and should be submit to ASoC list in addition to the devicetree list. Hence, add this information into the MAINTAINERS file. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 0abecc528dac..62701e0135af 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13930,6 +13930,7 @@ S: Supported F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/soc/ F: sound/soc/ +F: include/dt-bindings/sound/ F: include/sound/soc* SOUNDWIRE SUBSYSTEM -- 2.19.1

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[alsa-devel] [PATCH v2] ASoC: imx-audmux: complete dt-bindings for i.MX6
by Clément Péron 06 Dec '18

06 Dec '18

From: Colin Didier <colin.didier(a)devialet.com> The i.MX6 audmux differs from i.MX51. This patch adds the audmux for i.MX6 family. Signed-off-by: Colin Didier <colin.didier(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v2: Repost with ASoC maintainer and list include/dt-bindings/sound/fsl-imx-audmux.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 15f138bebe16..a1d0741d9ed1 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -25,6 +25,14 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +#define MX6_AUDMUX_PORT1_SSI1 0 +#define MX6_AUDMUX_PORT2_SSI2 1 +#define MX6_AUDMUX_PORT3 2 +#define MX6_AUDMUX_PORT4 3 +#define MX6_AUDMUX_PORT5 4 +#define MX6_AUDMUX_PORT6 5 +#define MX6_AUDMUX_PORT7_SSI3 6 + /* * TFCSEL/RFCSEL (i.MX27) or TFSEL/TCSEL/RFSEL/RCSEL (i.MX31/51/53/6Q) * can be sourced from Rx/Tx. -- 2.19.1

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