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[PATCH] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 22 Feb '23

22 Feb '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 66 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 7 +++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 75 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..e285db7 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,66 @@ +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + SectionDevice."Speaker".EnableSequence [ + cset "name='Playback Mux' HP" + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' SPK" + ] + + Value { + PlaybackMasterElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMasterElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..6e326af --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,7 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} + diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

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[PATCH 00/10] ASoC: Initial support for Cirrus Logic CS35L56
by Richard Fitzgerald 22 Feb '23

22 Feb '23

The CS35L56 is a high-performance boosted mono audio amplifier. Supported control interfaces are I2C, SPI or SoundWire. Supported audio interfaces are I2S/TDM or SoundWire. The CS35L56 has a self-booting firmware in ROM, with the ability to patch the firmware and/or apply tunings. Patches #1 to #7 add support to cs_dsp and wm_adsp for self-booting firmware and the ability to apply a .bin file without having to apply a .wmfw. Patch #8 adds the ASoC driver. Patches #9 and #10 add support for this to the Intel sof_sdw machine driver. Richard Fitzgerald (3): ASoC: cs35l56: Add driver for Cirrus Logic CS35L56 ASoC: Intel: sof_sdw: Add support for Cirrus Logic CS35L56 ASoC: Intel: soc-acpi: Add CS35L56 Soundwire to TGL Simon Trimmer (7): firmware: cs_dsp: Introduce no_core_startstop for self-booting DSPs ASoC: wm_adsp: Use no_core_startstop to prevent creating preload control firmware: cs_dsp: Support DSPs that don't require firmware download ASoC: wm_adsp: Support DSPs that don't require firmware download ASoC: wm_adsp: Expose the DSP boot work actions as wm_adsp_power_up() ASoC: wm_adsp: Add support for loading bin files without wmfw ASoC: wm_adsp: Simplify the logging of requested firmware files MAINTAINERS | 2 + drivers/firmware/cirrus/cs_dsp.c | 17 +- include/linux/firmware/cirrus/cs_dsp.h | 1 + include/sound/cs35l56.h | 266 +++ sound/soc/codecs/Kconfig | 34 + sound/soc/codecs/Makefile | 10 + sound/soc/codecs/cs35l56-i2c.c | 83 + sound/soc/codecs/cs35l56-sdw.c | 525 ++++++ sound/soc/codecs/cs35l56-shared.c | 387 +++++ sound/soc/codecs/cs35l56-spi.c | 81 + sound/soc/codecs/cs35l56.c | 1451 +++++++++++++++++ sound/soc/codecs/cs35l56.h | 77 + sound/soc/codecs/wm_adsp.c | 63 +- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/sof_sdw.c | 7 + sound/soc/intel/boards/sof_sdw_common.h | 7 + sound/soc/intel/boards/sof_sdw_cs35l56.c | 115 ++ .../intel/common/soc-acpi-intel-tgl-match.c | 57 + 20 files changed, 3172 insertions(+), 16 deletions(-) create mode 100644 include/sound/cs35l56.h create mode 100644 sound/soc/codecs/cs35l56-i2c.c create mode 100644 sound/soc/codecs/cs35l56-sdw.c create mode 100644 sound/soc/codecs/cs35l56-shared.c create mode 100644 sound/soc/codecs/cs35l56-spi.c create mode 100644 sound/soc/codecs/cs35l56.c create mode 100644 sound/soc/codecs/cs35l56.h create mode 100644 sound/soc/intel/boards/sof_sdw_cs35l56.c -- 2.30.2

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Re: Firewire Audio Interface
by Takashi Sakamoto 22 Feb '23

22 Feb '23

Hi, (C.C.ed to alsa-devel) On Tue, Feb 21, 2023 at 10:21:14AM -0500, mj@mj wrote: > Hello Takashi! > I came across your name when trying to find software to make a TC > Konnekt 6 work in OSX Big Sur. > TC has abandoned the Konnekt series. > I saw this Github entry that I think is from you? > [1]https://github.com/alsa-project/snd-firewire-ctl-services/blob/master/REA… > I wanted to ask…is this your project? and would it allow me to use the > Konnekt with OSX 64-bit? I maintain the repository. Linux kernel has driver stack for audio and music unit in IEEE 1394 bus. I maintain the driver stack as well. For example, you can see backport drivers from mainline Linux kernel: * https://github.com/takaswie/snd-firewire-improve/ The repository you found is supplemental one. It provides user space service program to configure DSP functions such as channel strip effects. Besides, they are available in Linux only. As long as I know, no one work to develop craft drivers in MacOS for your device. P.S. I think you use 'Desktop Konnekt 6' or 'Konnekt 8'. Takashi Sakamoto

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[PATCH V3 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 22 Feb '23

22 Feb '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1335 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 108 ++ 7 files changed, 1744 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

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[PATCH 0/2] ASoC: cppcheck fixups
by Kuninori Morimoto 22 Feb '23

22 Feb '23

Hi Mark Cc Takashi This patch-set fixes the issues which is reported by cppcheck. sound/pci/asihpi/hpi6205.c | 2 +- sound/pci/hda/patch_ca0132.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) It still reports about sound/pci/ice1712/wm8766.c (A) that there is a case that "val2" might be used with uninitialized. But I'm not sure what is the good solution for it. So, I can do is only Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> static int snd_wm8766_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wm8766 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; u16 val1, val2; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); else { val1 = wm->regs[wm->ctl[n].reg1] & wm->ctl[n].mask1; val1 >>= __ffs(wm->ctl[n].mask1); if (wm->ctl[n].flags & WM8766_FLAG_STEREO) { val2 = wm->regs[wm->ctl[n].reg2] & wm->ctl[n].mask2; val2 >>= __ffs(wm->ctl[n].mask2); if (wm->ctl[n].flags & WM8766_FLAG_VOL_UPDATE) val2 &= ~WM8766_VOL_UPDATE; } } if (wm->ctl[n].flags & WM8766_FLAG_INVERT) { val1 = wm->ctl[n].max - (val1 - wm->ctl[n].min); if (wm->ctl[n].flags & WM8766_FLAG_STEREO) (A) val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); } ^^^^ ucontrol->value.integer.value[0] = val1; if (wm->ctl[n].flags & WM8766_FLAG_STEREO) ucontrol->value.integer.value[1] = val2; return 0; } -- 2.25.1

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[PATCH v5 0/4] Add CS35L41 shared boost feature
by Lucas Tanure 21 Feb '23

21 Feb '23

Valve's Steam Deck uses CS35L41 in shared boost mode, where both speakers share the boost circuit. Add this support in the shared lib, but for now, shared boost is not supported in HDA systems as would require BIOS changes. Based on David Rhodes shared boost patches. Also, fix boost config overwriting in IRQ found in the review and do a small refactor of the code. Changes from V4: - Fix Document subject Changes from V3: - Fix wrong code sent - Fix ISO C90 mixed declarations and code Changes from V2: - Drop External boost without VSPK Documentation - Move Shared boost to use values 2 and 3 - Revert back to reg_sequence but reading the value first and only update the necessary bits - Fix bug found by Intel kernel Test Robot Changes from V1: - Fix Documentation patch subject - New patch for External boost without VSPK Documentation - New patch to fix boost IRQ overwriting issue - New patch to refactor IRQ release error code - reinit_completion on pcm_startup - fix DRE switch overwriting - return IRQ_HANDLED in PLL_LOCK case Lucas Tanure (4): ASoC: cs35l41: Only disable internal boost ASoC: cs35l41: Refactor error release code ALSA: cs35l41: Add shared boost feature ASoC: dt-bindings: cirrus,cs35l41: Document CS35l41 shared boost .../bindings/sound/cirrus,cs35l41.yaml | 10 +- include/sound/cs35l41.h | 13 +- sound/pci/hda/cs35l41_hda.c | 6 +- sound/soc/codecs/cs35l41-lib.c | 73 +++++++++- sound/soc/codecs/cs35l41.c | 125 +++++++++--------- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 157 insertions(+), 71 deletions(-) -- 2.39.1

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[PATCH v2 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 21 Feb '23

21 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 ++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 55 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 +++++++-- 4 files changed, 80 insertions(+), 10 deletions(-) -- 2.34.1

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alsa-devel mailing list - mailman upgrade
by Jaroslav Kysela 21 Feb '23

21 Feb '23

Hi all, I upgraded the mailman 2 to mailman 3 on the ALSA server. I hope that you do not see any differences, but there is a major change in the handling of the user logins to maintain your subscription using the web interface [1]. It means that you should create a new user account on the mailman server [2]. Do not request the password change for the first login. Although the subscriptions are migrated, the new system will send you a reply that your e-mail is not in the user database. Although it is technically correct, it is very confusing. Create a new account using identical e-mail as you are subscribed. Your subscription will be assigned to this account according your e-mail. You will be allowed to maintain all list subscriptions on the server then (not only one list subscription like before). Archive URLs are different, but I will keep the old archive (URLs), too. Enjoy new Postorius and HyperKitty frontends. If you have any issue, please, let me know. Thanks, Jaroslav [1] https://docs.mailman3.org/en/latest/faq-migration.html [2] https://mailman.alsa-project.org -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH RFC 0/3] Add Versa3 clock generator support
by Biju Das 20 Feb '23

20 Feb '23

The 5P35023 is a VersaClock programmable clock generator and it provides 6 clk outputs {diff2, diff1, se3, se2, se1 and refin}. It has an internal OTP memory allows the user to store the configuration in the device. After power up, the user can change the device register settings through the I2C interface when I2C mode is selected. This driver is for overriding OTP default values during boot based on a full register map from DT, and also minimal support to change the parent of a output clock. The motivation for developing this driver is for supporting 48KHz playback/record with audio codec on RZ/G2L SMARC EVK. On RZ/G2L SMARC EVK, By default audio mclk is connected to 11.2896 MHz clk which is multiple of 44.1KHz. Please see the below default OTP configuration of Dividers connected to output clocks. DIV3 12.2880 MHz DIFF2--> Audio clk2 DIV5 11.2896 MHz SE1 --> Audio clk1 DIV5 11.2896 MHz SE2 --> Audio mck DIV4 12 MHz SE3 --> This clk Not used DIV1 25 MHz DIFF1-->Ethernet clk Ref1-> 24MHz With this setup, we won't be able to do 48KHz playback/record on audio codec, as mck is always connected to 11.2896MHz clk. But by programming the i2c, we can make use of DIV4 to generate 12.2880 MHz and make that as parent of SE2 and there by supporting 48KHz playback/record. A block diagram with modification can be find here[1] [1]https://paste.pics/a253ce7cdc8720c3b5eb6953b97b25ff DIV3 12.2880 MHz DIFF2--> Audio clk2 DIV5 11.2896 MHz SE1 --> Audio clk1 DIV5 11.2896 MHz | SE2 --> Audio mck DIV4 12.2880 MHz | DIV2 12 MHz SE3 --> This clk Not used DIV1 25 MHz DIFF1--> Ethernet clk Ref1-> 24MHz The driver can read a full register map from the DT, and will use that register map to initialize the clk generator when the system boots. and later, based on sampling rate, it switches the parent of SE2 and support both 44.1 and 48 KHz playback/record at run time. 48KHz playback 1f: f6 --> setting Div4 as clock source for se2 Read at address 0x10049C00 : 0x300B4022 --> Setting Audio clk2 in SSI pfd2 1 1 0 24000000 pll2 1 1 0 491519897 div4_mux 1 1 0 491519897 div4 1 1 0 12287998 se2_mux 1 1 0 12287998 se2 1 1 0 12287998 44.1KHz playback 1f: b6 --> setting Div5 as clock source for se2 Read at address 0x10049C00: 0x700B4022--> Setting Audio clk1 in SSI pfd3_mux 1 1 0 24000000 pfd3 1 1 0 960000 pll3 1 1 0 564480000 div5 1 1 0 11289600 se2_mux 1 1 0 11289600 se2 1 1 0 11289600 Please provide your valuable comment for this patch series. Biju Das (3): dt-bindings: clock: Add Renesas versa3 clock generator bindings drivers: clk: Add support for versa3 clock driver arm64: dts: renesas: rzg2l-smarc: Use versa3 clk for audio mclk .../bindings/clock/renesas,versaclock3.yaml | 135 ++ .../boot/dts/renesas/rz-smarc-common.dtsi | 7 - arch/arm64/boot/dts/renesas/rzg2l-smarc.dtsi | 35 + drivers/clk/Kconfig | 9 + drivers/clk/Makefile | 1 + drivers/clk/clk-versaclock3.c | 1134 +++++++++++++++++ 6 files changed, 1314 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/renesas,versaclock3.yaml create mode 100644 drivers/clk/clk-versaclock3.c -- 2.25.1

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[PATCH BACKPORT 5.4] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 20 Feb '23

20 Feb '23

[ Upstream commit 1f810d2b6b2fbdc5279644d8b2c140b1f7c9d43d ] There were just too many code changes since 5.4 stable to prevent clean picking the fix from mainline. The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Cc: stable(a)vger.kernel.org # 5.4.x Complements: b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230216162340.19480-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index b3cdd10c83ae..c30e450fa970 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -211,6 +211,10 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, int stream_tag; int ret; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + /* get stored dma data if resuming from system suspend */ link_dev = snd_soc_dai_get_dma_data(dai, substream); if (!link_dev) { @@ -231,10 +235,6 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - /* set the stream tag in the codec dai dma params */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); -- 2.39.2

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