September 2023 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

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[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

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[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 17 Jun '25

17 Jun '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

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[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

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ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

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hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

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[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

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[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

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[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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[PATCH v4 0/7] Patches to update for rockchip pdm
by Sugar Zhang 23 Apr '24

23 Apr '24

These patches fixup or update for rockchip pdm. Changes in v4: - Acked by Rob Herring Changes in v3: - Fix property 'path-map' suggested by Rob Herring. Changes in v2: - Fix yamllint errors. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 -------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 120 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 +++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 232 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

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[PATCH v3] powerpc: macio: Make remove callback of macio driver void returned
by Dawei Li 13 Mar '24

13 Mar '24

Commit fc7a6209d571 ("bus: Make remove callback return void") forces bus_type::remove be void-returned, it doesn't make much sense for any bus based driver implementing remove callbalk to return non-void to its caller. This change is for macio bus based drivers. Signed-off-by: Dawei Li <set_pte_at(a)outlook.com> --- v2 -> v3 - Rebased on latest powerpc/next. - cc' to relevant subsysem lists. v1 -> v2 - Revert unneeded changes. - Rebased on latest powerpc/next. v1 - https://lore.kernel.org/all/TYCP286MB2323FCDC7ECD87F8D97CB74BCA189@TYCP286M… --- arch/powerpc/include/asm/macio.h | 2 +- drivers/ata/pata_macio.c | 4 +--- drivers/macintosh/rack-meter.c | 4 +--- drivers/net/ethernet/apple/bmac.c | 4 +--- drivers/net/ethernet/apple/mace.c | 4 +--- drivers/net/wireless/intersil/orinoco/airport.c | 4 +--- drivers/scsi/mac53c94.c | 5 +---- drivers/scsi/mesh.c | 5 +---- drivers/tty/serial/pmac_zilog.c | 7 ++----- sound/aoa/soundbus/i2sbus/core.c | 4 +--- 10 files changed, 11 insertions(+), 32 deletions(-) diff --git a/arch/powerpc/include/asm/macio.h b/arch/powerpc/include/asm/macio.h index ff5fd82d9ff0..cb9c386dacf8 100644 --- a/arch/powerpc/include/asm/macio.h +++ b/arch/powerpc/include/asm/macio.h @@ -125,7 +125,7 @@ static inline struct pci_dev *macio_get_pci_dev(struct macio_dev *mdev) struct macio_driver { int (*probe)(struct macio_dev* dev, const struct of_device_id *match); - int (*remove)(struct macio_dev* dev); + void (*remove)(struct macio_dev *dev); int (*suspend)(struct macio_dev* dev, pm_message_t state); int (*resume)(struct macio_dev* dev); diff --git a/drivers/ata/pata_macio.c b/drivers/ata/pata_macio.c index 9ccaac9e2bc3..653106716a4b 100644 --- a/drivers/ata/pata_macio.c +++ b/drivers/ata/pata_macio.c @@ -1187,7 +1187,7 @@ static int pata_macio_attach(struct macio_dev *mdev, return rc; } -static int pata_macio_detach(struct macio_dev *mdev) +static void pata_macio_detach(struct macio_dev *mdev) { struct ata_host *host = macio_get_drvdata(mdev); struct pata_macio_priv *priv = host->private_data; @@ -1202,8 +1202,6 @@ static int pata_macio_detach(struct macio_dev *mdev) ata_host_detach(host); unlock_media_bay(priv->mdev->media_bay); - - return 0; } #ifdef CONFIG_PM_SLEEP diff --git a/drivers/macintosh/rack-meter.c b/drivers/macintosh/rack-meter.c index c28893e41a8b..f2f83c4f3af5 100644 --- a/drivers/macintosh/rack-meter.c +++ b/drivers/macintosh/rack-meter.c @@ -523,7 +523,7 @@ static int rackmeter_probe(struct macio_dev* mdev, return rc; } -static int rackmeter_remove(struct macio_dev* mdev) +static void rackmeter_remove(struct macio_dev *mdev) { struct rackmeter *rm = dev_get_drvdata(&mdev->ofdev.dev); @@ -558,8 +558,6 @@ static int rackmeter_remove(struct macio_dev* mdev) /* Get rid of me */ kfree(rm); - - return 0; } static int rackmeter_shutdown(struct macio_dev* mdev) diff --git a/drivers/net/ethernet/apple/bmac.c b/drivers/net/ethernet/apple/bmac.c index 9e653e2925f7..292b1f9cd9e7 100644 --- a/drivers/net/ethernet/apple/bmac.c +++ b/drivers/net/ethernet/apple/bmac.c @@ -1591,7 +1591,7 @@ bmac_proc_info(char *buffer, char **start, off_t offset, int length) } #endif -static int bmac_remove(struct macio_dev *mdev) +static void bmac_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct bmac_data *bp = netdev_priv(dev); @@ -1609,8 +1609,6 @@ static int bmac_remove(struct macio_dev *mdev) macio_release_resources(mdev); free_netdev(dev); - - return 0; } static const struct of_device_id bmac_match[] = diff --git a/drivers/net/ethernet/apple/mace.c b/drivers/net/ethernet/apple/mace.c index fd1b008b7208..e6350971c707 100644 --- a/drivers/net/ethernet/apple/mace.c +++ b/drivers/net/ethernet/apple/mace.c @@ -272,7 +272,7 @@ static int mace_probe(struct macio_dev *mdev, const struct of_device_id *match) return rc; } -static int mace_remove(struct macio_dev *mdev) +static void mace_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct mace_data *mp; @@ -296,8 +296,6 @@ static int mace_remove(struct macio_dev *mdev) free_netdev(dev); macio_release_resources(mdev); - - return 0; } static void dbdma_reset(volatile struct dbdma_regs __iomem *dma) diff --git a/drivers/net/wireless/intersil/orinoco/airport.c b/drivers/net/wireless/intersil/orinoco/airport.c index a890bfa0d5cc..276a06cdd1f5 100644 --- a/drivers/net/wireless/intersil/orinoco/airport.c +++ b/drivers/net/wireless/intersil/orinoco/airport.c @@ -85,7 +85,7 @@ airport_resume(struct macio_dev *mdev) return err; } -static int +static void airport_detach(struct macio_dev *mdev) { struct orinoco_private *priv = dev_get_drvdata(&mdev->ofdev.dev); @@ -111,8 +111,6 @@ airport_detach(struct macio_dev *mdev) macio_set_drvdata(mdev, NULL); free_orinocodev(priv); - - return 0; } static int airport_hard_reset(struct orinoco_private *priv) diff --git a/drivers/scsi/mac53c94.c b/drivers/scsi/mac53c94.c index f75928f7773e..42648ca9b8ed 100644 --- a/drivers/scsi/mac53c94.c +++ b/drivers/scsi/mac53c94.c @@ -508,7 +508,7 @@ static int mac53c94_probe(struct macio_dev *mdev, const struct of_device_id *mat return rc; } -static int mac53c94_remove(struct macio_dev *mdev) +static void mac53c94_remove(struct macio_dev *mdev) { struct fsc_state *fp = (struct fsc_state *)macio_get_drvdata(mdev); struct Scsi_Host *host = fp->host; @@ -526,11 +526,8 @@ static int mac53c94_remove(struct macio_dev *mdev) scsi_host_put(host); macio_release_resources(mdev); - - return 0; } - static struct of_device_id mac53c94_match[] = { { diff --git a/drivers/scsi/mesh.c b/drivers/scsi/mesh.c index 84b541a57b7b..cd2575b88c85 100644 --- a/drivers/scsi/mesh.c +++ b/drivers/scsi/mesh.c @@ -1986,7 +1986,7 @@ static int mesh_probe(struct macio_dev *mdev, const struct of_device_id *match) return -ENODEV; } -static int mesh_remove(struct macio_dev *mdev) +static void mesh_remove(struct macio_dev *mdev) { struct mesh_state *ms = (struct mesh_state *)macio_get_drvdata(mdev); struct Scsi_Host *mesh_host = ms->host; @@ -2013,11 +2013,8 @@ static int mesh_remove(struct macio_dev *mdev) macio_release_resources(mdev); scsi_host_put(mesh_host); - - return 0; } - static struct of_device_id mesh_match[] = { { diff --git a/drivers/tty/serial/pmac_zilog.c b/drivers/tty/serial/pmac_zilog.c index 13668ffdb1e7..d4640479c338 100644 --- a/drivers/tty/serial/pmac_zilog.c +++ b/drivers/tty/serial/pmac_zilog.c @@ -1507,12 +1507,12 @@ static int pmz_attach(struct macio_dev *mdev, const struct of_device_id *match) * That one should not be called, macio isn't really a hotswap device, * we don't expect one of those serial ports to go away... */ -static int pmz_detach(struct macio_dev *mdev) +static void pmz_detach(struct macio_dev *mdev) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); if (!uap) - return -ENODEV; + return; uart_remove_one_port(&pmz_uart_reg, &uap->port); @@ -1523,11 +1523,8 @@ static int pmz_detach(struct macio_dev *mdev) dev_set_drvdata(&mdev->ofdev.dev, NULL); uap->dev = NULL; uap->port.dev = NULL; - - return 0; } - static int pmz_suspend(struct macio_dev *mdev, pm_message_t pm_state) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 51ed2f34b276..35f39727994d 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -364,15 +364,13 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return 0; } -static int i2sbus_remove(struct macio_dev* dev) +static void i2sbus_remove(struct macio_dev *dev) { struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) soundbus_remove_one(&i2sdev->sound); - - return 0; } #ifdef CONFIG_PM -- 2.25.1

6 6

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Amadeusz Sławiński 03 Mar '24

03 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..652aae4e221d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1717,7 +1717,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1770,7 +1770,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1814,7 +1814,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.34.1

2 2

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Michele Perrone 29 Feb '24

29 Feb '24

Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 120 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 72 +++++++++++++++++++ 3 files changed, 193 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..ad22bfafd324 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,120 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-weiss.c - a part of driver for DICE based devices + * + * Copyright (c) 2023 Rolf Anderegg and Michele Perrone + */ + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +/* Weiss DAC202: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss MAN301: 192kHz 2-channel music archive network player */ +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{0, 0, 0}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .has_midi = false +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, /* Maya edition: same audio I/O as DAC202 */ + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..4bcd026a0b3f 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -396,6 +396,78 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* Weiss DAC202: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC202: 192kHz 2-channel DAC (Maya edition) */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss MAN301: 192kHz 2-channel music archive network player */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); base-commit: 9b4469410cf9a0fcbccc92c480fd42f7c815a745 -- 2.41.0

3 26

Codec stops working after sleep and resume
by Jesús Jiménez Sánchez 15 Feb '24

15 Feb '24

I am using the tlv320aic31xx codec in my embedded device and I've noticed that, after forcing the unit into sleep and then, resume, I get a bunch of these errors: ``` tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: aic31xx_wait_bits: Failed! 0x25 was 0xcc239ca8 expected 0x0 (-108, 0x20, 0 us) tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: HPL Driver event failed: -1 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPL ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPR ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 ``` When I look at he output of `dmesg`, I found this: ``` ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1506 at drivers/i2c/i2c-core.h:54 __i2c_transfer+0xac/0xb4 i2c i2c-0: Transfer while suspended Modules linked in: rfcomm stm32_adc stm32_timer_trigger stm32_lptimer_trigger sha256_generic libsha256 sha256_arm snd_soc_simple_card snd_soc_simple_card_utils cfg80211 stm32_crc32 hci_uart btqca stm32_cryp libdes crypto_engine snd_soc_stm32_i2s snd_soc_tlv320aic31xx bluetooth stm32_adc_core snd_soc_core atmel_mxt_ts snd_pcm_dmaengine snd_pcm ecdh_generic snd_timer snd ecc sgm4151x_charger soundcore libaes [last unloaded: wlan] CPU: 0 PID: 1506 Comm: sh Tainted: G O 5.15.67 #1 Hardware name: STM32 (Device Tree Support) [<c010e8cc>] (unwind_backtrace) from [<c010c120>] (show_stack+0x10/0x14) [<c010c120>] (show_stack) from [<c011dc1c>] (__warn+0xec/0x148) [<c011dc1c>] (__warn) from [<c0bed42c>] (warn_slowpath_fmt+0x78/0xac) [<c0bed42c>] (warn_slowpath_fmt) from [<c092ecdc>] (__i2c_transfer+0xac/0xb4) [<c092ecdc>] (__i2c_transfer) from [<c092ed80>] (i2c_transfer+0x9c/0x138) [<c092ed80>] (i2c_transfer) from [<c06f5a40>] (regmap_i2c_read+0x5c/0x98) [<c06f5a40>] (regmap_i2c_read) from [<c06f0e2c>] (_regmap_raw_read+0x10c/0x2a8) [<c06f0e2c>] (_regmap_raw_read) from [<c06f1008>] (_regmap_bus_read+0x40/0x6c) [<c06f1008>] (_regmap_bus_read) from [<c06ef9c4>] (_regmap_read+0x60/0x14c) [<c06ef9c4>] (_regmap_read) from [<c06effdc>] (_regmap_update_bits+0xb4/0xf4) [<c06effdc>] (_regmap_update_bits) from [<c06f00c0>] (_regmap_select_page+0xa4/0x108) [<c06f00c0>] (_regmap_select_page) from [<c06f07a8>] (_regmap_raw_write_impl+0x684/0x968) [<c06f07a8>] (_regmap_raw_write_impl) from [<c06f0b04>] (_regmap_bus_raw_write+0x78/0xa0) [<c06f0b04>] (_regmap_bus_raw_write) from [<c06efe40>] (_regmap_write+0x4c/0x134) [<c06efe40>] (_regmap_write) from [<c06f0010>] (_regmap_update_bits+0xe8/0xf4) [<c06f0010>] (_regmap_update_bits) from [<c06f1820>] (regmap_update_bits_base+0x50/0x74) [<c06f1820>] (regmap_update_bits_base) from [<bf0d0980>] (snd_soc_component_update_bits+0x78/0x124 [snd_soc_core]) [<bf0d0980>] (snd_soc_component_update_bits [snd_soc_core]) from [<bf0daa20>] (dapm_seq_run_coalesced+0x14c/0x1a4 [snd_soc_core]) [<bf0daa20>] (dapm_seq_run_coalesced [snd_soc_core]) from [<bf0dab68>] (dapm_seq_run+0xf0/0x350 [snd_soc_core]) [<bf0dab68>] (dapm_seq_run [snd_soc_core]) from [<bf0db170>] (dapm_power_widgets+0x3a8/0x530 [snd_soc_core]) [<bf0db170>] (dapm_power_widgets [snd_soc_core]) from [<bf0cc214>] (snd_soc_dapm_sync+0x48/0x54 [snd_soc_core]) [<bf0cc214>] (snd_soc_dapm_sync [snd_soc_core]) from [<bf0c652c>] (snd_soc_suspend+0xf0/0x230 [snd_soc_core]) [<bf0c652c>] (snd_soc_suspend [snd_soc_core]) from [<c06df188>] (dpm_run_callback+0x48/0x134) [<c06df188>] (dpm_run_callback) from [<c06df58c>] (__device_suspend+0x11c/0x4c4) [<c06df58c>] (__device_suspend) from [<c06e1c10>] (dpm_suspend+0x148/0x2cc) [<c06e1c10>] (dpm_suspend) from [<c06e20e8>] (dpm_suspend_start+0x64/0x6c) [<c06e20e8>] (dpm_suspend_start) from [<c0159f00>] (suspend_devices_and_enter+0x150/0x2b4) [<c0159f00>] (suspend_devices_and_enter) from [<c0bede38>] (enter_state+0x204/0x250) [<c0bede38>] (enter_state) from [<c015a0a4>] (pm_suspend+0x40/0x98) [<c015a0a4>] (pm_suspend) from [<c0158ae4>] (state_store+0x68/0xc8) [<c0158ae4>] (state_store) from [<c0330a68>] (kernfs_fop_write_iter+0x10c/0x1cc) [<c0330a68>] (kernfs_fop_write_iter) from [<c028ea40>] (vfs_write+0x298/0x45c) [<c028ea40>] (vfs_write) from [<c028ed4c>] (ksys_write+0x60/0xe4) [<c028ed4c>] (ksys_write) from [<c0100060>] (ret_fast_syscall+0x0/0x48) Exception stack(0xcffcbfa8 to 0xcffcbff0) bfa0: 00000004 005ac7d8 00000001 005ac7d8 00000004 00000000 bfc0: 00000004 005ac7d8 00000001 00000004 b6f01680 bee896d8 005a8448 005a8590 bfe0: 005a6dfc bee89690 004fcda4 b6dcb4f8 ---[ end trace 75457fd792755cae ]--- ``` I've been trying for a few days, but I can't find the source of the problem or a way to solve it. Is there anything I might be missing? I assume the problem is that it's trying to write to the registers after the suspend has happened, but I haven't been able to prove that either, and I wouldn't know how to fix that. Could that be what's happening here? Thanks.

1 1

[PATCH v7 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 Jan '24

31 Jan '24

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. This series is mostly just a resend keeping pace with the kernel under it, except for a minor fixup in the ASoC stuff. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1188 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 284 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 946 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

6 32

[v3 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 31 Jan '24

31 Jan '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 32 +++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 113 insertions(+), 10 deletions(-) -- 2.25.1

5 13

[RESEND PATCH 0/1] ALSA: virtio: add support for audio controls
by Aiswarya Cyriac 15 Jan '24

15 Jan '24

From: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Implementation of support for audio controls in accordance with the extension of the virtio sound device specification[1] planned for virtio-v1.3-cs01. Design　of virtual audio controls is based on and derived from ALSA audio controls. It allows the driver to perform all standard operations, such as reading and writing audio control value, as well as working with metadata (represented in the TLV form). The driver part was tested on top of the Linux 5.10 kernel. As a device part was used OpenSynergy proprietary implementation. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Anton Yakovlev (1): ALSA: virtio: add support for audio controls include/uapi/linux/virtio_snd.h | 154 +++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 464 ++++++++++++++++++++++++++++++++ 5 files changed, 662 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c -- 2.25.1

2 3

[PATCH 00/31] Fix and improve Rockchip RK3128 support
by Alex Bee 13 Dec '23

13 Dec '23

Hi list, this series fixes some issues I found when testing my "new" RK3128 board with the mainline kernel and adds some core functionality like SMP bringup, usb and networking. The propably most distinctive change is the split up of the DTs for the different SoCs of this platform: RK3126 and RK3128. Even if I'm not adding a RK3126 board in this series: I think this change should be done as early as possible in order to avoid issues in future. Actually it should have been done like that in the first place. All patches have been tested on both SoCs: Patch 01-04 are dt-bindings additions for this SoC platform and the new board I'm addding here Patch 05-07 are some clock driver fixes (there is more to do, but I'm limiting it to the things which can work now) Patch 08 and 09 adding support for the usb 2 phy found in RK312x platform to the Rockchip Innosilicon usb phy driver. Patch 10-14 are DT patches which I did in the "old" rk3128.dtsi in order to be backportable. Patch 15 does the aforementioned split-up of the current rk3128.dtsi in rk312x.dtsi, rk3126.dtsi and rk3128.dtsi Patch 16-20 adds SMP bringup and cpu frequency scaling Patch 21-23 adds power controller, GPU and I2S nodes to the RK312x DT Patch 24-26 adds the second I2S, S/PDIF and gmac nodes to the RK3128 DT Patch 27-30 adds fixes for the usb controllers and adds a missing pincontrol for the sd card detection Patch 31 finnaly adds a DT for Genatech XPI-3128 board There are quite some HW blocks which can easily be added for this platform, but I guess this series is pretty huge already, so that it is limited to some core devices for now. I hope the single patches are small enough, that reviewing won't be a burden ;) The plan is to sync the DT changes to u-boot after they are merged where I'm planning to add "full" TPL/SPL for this platform (dram controller/phy driver is currently missing) I'm planing the add a RK3126 (tablet) board here as well, but I will have to add support for RK816 pmic first. Alex Bee (29): dt-bindings: mfd: syscon: Add rockchip,rk3128-qos compatible dt-bindings: gpu: mali-utgard: Add Rockchip RK3128 compatible dt-bindings: ASoC: rockchip: Add compatible for RK3128 spdif dt-bindings: arm: rockchip: Add Geniatech XPI-3128 clk: rockchip: rk3128: Fix SCLK_SDMMC's clock name phy: rockchip-inno-usb2: Split ID interrupt phy registers phy: phy-rockchip-inno-usb2: Add RK3128 support ARM: dts: rockchip: Fix i2c0 register address for RK3128 ARM: dts: rockchip: Add missing arm timer interrupt for RK3128 ARM: dts: rockchip: Add missing quirk for RK3128's dma engine ARM: dts: rockchip: Fix timer clocks for RK3128 ARM: dts: rockchip: Disable non-required timers for RK3128 ARM: dts: rockchip: Split RK3128 devictree for RK312x SoC family ARM: dts: rockchip: Add SRAM node for RK312x ARM: dts: rockchip: Add CPU resets for RK312x ARM: dts: rockchip: Enable SMP bringup for RK312x ARM: dts: rockchip: Switch to operating-points-v2 for RK312x's CPU ARM: dts: rockchip: Add extra CPU voltages for RK3126 ARM: dts: rockchip: add power controller for RK312x ARM: dts: rockchip: Add GPU node for RK312x ARM: dts: rockchip: Add 2-channel I2S for RK312x ARM: dts: rockchip: Add 8-channel I2S for RK3128 ARM: dts: rockchip: Add spdif for RK3128 ARM: dts: rockchip: Add gmac for RK3128 ARM: dts: rockchip: Add dwc2 otg fifo siztes for RK312x ARM: dts: rockchip: Add USB host clocks for RK312x ARM: dts: rockchip: Make usbphy the parent of SCLK_USB480M for RK312x ARM: dts: rockchip: Add sdmmc_det pinctrl for RK312x ARM: dts: Add Geniatech XPI-3128 RK3128 board Finley Xiao (2): clk: rockchip: rk3128: Fix aclk_peri_src parent clk: rockchip: rk3128: Fix hclk_otg gate .../devicetree/bindings/arm/rockchip.yaml | 5 + .../bindings/gpu/arm,mali-utgard.yaml | 1 + .../devicetree/bindings/mfd/syscon.yaml | 1 + .../bindings/sound/rockchip-spdif.yaml | 1 + arch/arm/boot/dts/rockchip/Makefile | 1 + arch/arm/boot/dts/rockchip/rk3126.dtsi | 36 + .../arm/boot/dts/rockchip/rk3128-xpi-3128.dts | 431 +++++++ arch/arm/boot/dts/rockchip/rk3128.dtsi | 920 +------------- arch/arm/boot/dts/rockchip/rk312x.dtsi | 1101 +++++++++++++++++ drivers/clk/rockchip/clk-rk3128.c | 24 +- drivers/phy/rockchip/phy-rockchip-inno-usb2.c | 147 ++- 11 files changed, 1754 insertions(+), 914 deletions(-) create mode 100644 arch/arm/boot/dts/rockchip/rk3126.dtsi create mode 100644 arch/arm/boot/dts/rockchip/rk3128-xpi-3128.dts create mode 100644 arch/arm/boot/dts/rockchip/rk312x.dtsi base-commit: 2dde18cd1d8fac735875f2e4987f11817cc0bc2c -- 2.42.0

6 46

Regression apparently caused by commit 088a40980efbc2c449b72f0f2c7ebd82f71d08e2 "ASoC: amd: acp: add pm ops support for acp pci driver"
by Marian Postevca 05 Dec '23

05 Dec '23

I'm trying to develop a sound machine driver based on the acp legacy driver. The first version of the driver was sent for review on the alsa mailing list this spring: https://lore.kernel.org/all/20230320203519.20137-1-posteuca@mutex.one I'm trying to fix some of the issues that were brought up during the review back then, but when I ported the patches to the latest commit on the for-next branch, I noticed a regression where I couldn't hear any sound at all. So I started a bisect session and found that the first bad commit is: ASoC: amd: acp: add pm ops support for acp pci driver commit 088a40980efbc2c449b72f0f2c7ebd82f71d08e2 https://lore.kernel.org/lkml/20230622152406.3709231-11-Syed.SabaKareem@amd.… If I revert this commit sound works as expected. So I started tinkering a little bit with it and I believe that what happens is that the acp pci driver enters the autosuspend state and never leaves this state at all. I noticed this because if I increase the autosuspend delay to a much larger value, then the sound works until that delay passes. I added traces and I can see that when the delay expires the suspend callback snd_acp_suspend() gets called, but the resume callback snd_acp_resume() never gets called. I'm no expert in runtime power management (though I did read a bit on it), so I don't understand all the things that happen underneath, but one thing that is not clear to me is who's supposed to mark activity on this device and keep it from entering autosuspend if the user wants to play some sound? Shouldn't there be some counterpart that calls pm_runtime_mark_last_busy() ? I looked through the code and can't find who's calling pm_runtime_mark_last_busy(). Some help here would be welcome. Is there something missing in my machine driver code, or is the runtime pm handling in acp pci driver wrong?

5 12

Yamaha YIT-W12TX not working
by Julian Sikorski 27 Nov '23

27 Nov '23

Hi list, Yamaha YIT-W12TX is a USB dongle which allows to transmit audio from PC to the YSP-4300 sound bar [1][2]. While it is quite old and no longer sold, I recently tried to make it work under linux. Unfortunately, it did not work: [ 3676.366404] usb 2-1.1: new full-speed USB device number 5 using ehci-pci [ 3676.445995] usb 2-1.1: config 1 has an invalid interface number: 4 but max is 3 [ 3676.446001] usb 2-1.1: config 1 has no interface number 2 [ 3676.446664] usb 2-1.1: New USB device found, idVendor=0499, idProduct=3108, bcdDevice= 1.00 [ 3676.446666] usb 2-1.1: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [ 3676.446679] usb 2-1.1: Product: YIT-W12TX [ 3676.446683] usb 2-1.1: Manufacturer: YAMAHA Corp. [ 3676.449508] hid-generic 0003:0499:3108.0001: hiddev96,hidraw0: USB HID v0.01 Device [YAMAHA Corp. YIT-W12TX] on usb-0000:00:1d.0-1.1/input3 [ 3676.450846] input: YAMAHA Corp. YIT-W12TX as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.1/2-1.1:1.4/0003:0499:3108.0002/input/input34 [ 3676.503844] hid-generic 0003:0499:3108.0002: input,hidraw1: USB HID v1.00 Device [YAMAHA Corp. YIT-W12TX] on usb-0000:00:1d.0-1.1/input4 [ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1 [ 3676.675786] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.676476] usbcore: registered new interface driver snd-usb-audio [ 3676.731911] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.735658] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.739399] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.743131] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.746920] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.750648] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.754445] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.758524] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.762274] usb 2-1.1: 5:0: cannot get min/max values for control 2 (id 5) [ 3676.789671] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790049] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790414] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.790773] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.791169] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792134] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792547] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.792899] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.793254] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.793634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.794900] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.795255] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.795634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.796008] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.796384] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.797543] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.797879] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.798254] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.798634] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.799007] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.813819] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.814281] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.814686] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.815281] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.815929] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.817807] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.818155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.818592] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.819165] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.819801] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.822163] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.822802] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.823284] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.824161] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.824907] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.827683] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.828159] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.828675] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.829415] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.830157] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.841801] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.842155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.842551] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.843158] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.843785] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.844909] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.845283] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.845673] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.846156] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.846800] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.848287] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.848799] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.849280] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.850031] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.850799] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.852677] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.853154] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.853674] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.854429] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.855155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.858916] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.859277] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.859672] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.860280] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.860909] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862033] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862418] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.862797] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.863279] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.863923] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.865471] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.866033] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.866590] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.867410] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.868157] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.870162] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.870674] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.871155] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.871908] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.872676] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 [ 3676.900672] usb 2-1.1: USB disconnect, device number 5 [ 3676.900688] usb 2-1.1: 1:1: cannot set freq 44100 to ep 0x1 This is with kernel-5.9.13-200.fc33.x86_64 and alsa-lib-1.2.4-5.fc33.x86_64 on Fedora 33. Under Windows 10 the dongle works without any special drivers from Yamaha, everything appears to be downloaded by Windows automatically, so I was hoping the device could be made to work under Linux relatively easily. Best regards, Julian [1] https://europe.yamaha.com/en/products/audio_visual/accessories/yit-w12/down… [2] https://europe.yamaha.com/en/products/audio_visual/sound_bar/ysp-4300/downl…

2 7

[PATCH v2 0/1] ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive
by Chancel Liu 27 Nov '23

27 Nov '23

It's reasonable that DAI parameters should be cleared if current DAI becomes inactive. Only check DAI active status in soc_pcm_hw_free() is not enough since there's the risk that DAI parameters never be cleared if there're more than one stream [see A]. Only check DAI active status in soc_pcm_close() is also not enough since it will cause the cleanup just happening in soc_pcm_close() [see B]. [A] For example, we have stream1 and stream2 for DAI about to stop. stream2 executes soc_pcm_hw_free() before stream1 executes soc_pcm_close(). At the moment, stream2 should clear current DAI parameters because stream1 passed the cleanup stage in soc_pcm_hw_free() and stream2 in fact is the only active stream which has a chance to do clean up. Since DAI active status is not yet updated by stream1 in soc_pcm_close(), stream2 doesn't know itself should clear DAI parameters. In result both stream1 and stream2 don't clear the parameters. [B] Suppose a case: aplay -Dhw:0 44100.wav 48000.wav In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free() ->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. The DAI parameters would be remained in the system even if the playback of 44100.wav is finished. In conclusion, it's better to check DAI active status in both soc_pcm_hw_free() and soc_pcm_close() which makes sure DAI parameters cleared if the DAI becomes inactive. changes in v2: - Patch v1 link: https://lore.kernel.org/all/20230112065834.580192-1-chancel.liu@nxp.com/ Patch v1 tries to introduce a usage count called hw_params_count to fix issue on DAI parameters cleanup. However it's not a good fix because not considering hw_params() and hw_free() are not symmetrical and hw_params() might be called multilpe times by user. - Both check DAI active status in soc_pcm_hw_free() and soc_pcm_close() which makes sure DAI parameters cleared if the DAI becomes inactive. Chancel Liu (1): ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive sound/soc/soc-pcm.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) -- 2.25.1

3 4

[PATCH v2 00/17] ALSA/ASoC: hda: Address format selection limitations and ambiguity
by Cezary Rojewski 14 Nov '23

14 Nov '23

Patchset aims to address format selection restrictions present currently in the HDAudio library. Formats which we are concerned about are 20 and 24 valid bits per sample within 32 bit depth container. One may identify them as S20_LE and S24_LE except that those, according to comments found in include/uapi/sound/asound.h, are for LSB-aligned scenarios. HDAudio streams expect MSB-aligned data, no matter if we are speaking of HOST (SDxFMT) or LINK (PPLCxFMT) side - chapter 4.5.1 of the public HDAudio specification. In short, S20_LE and S24_LE are invalid options. Right now, given the implementation of snd_hdac_query_supported_pcm() within sound/hda/hdac_device.c, even if a codec responds with: "I support all possible formats specified within HDAudio specification", there will be no option to open a 20/32 or 24/32 stream. The kernel will force the stream to be opened using the highest available bit depth. After discussing subject initially with Jaroslav and Takashi, suggestion was made to utilize 'subformat' option to address the problem. The eye-opening discussion begun much earlier though, in 2019 [1]. Paired with PRs for alsa-utils [2] and alsa-lib [3]. Flow of changes: The very first patch adds MSBITS subformat options to allow for granular 20/32, 24/32 and 32/32 format selection. The next three make sure subformat is actually honored during runtime. Most of that code is based on format-related API. Follow up is upgrade to the hda stream-format interface - several functions are added to make the granular format selection simple in the HDAudio world. Core of the implementation is based on the existing snd_hdac_calc_stream_format(). The next ten patches are straightforward switch from one interface to another with cleanup of now-unsed function as a finishing touch. Last but not least - the avs-driver, on which the problem analyzed and debugged, is updated to no longer acknowledge S24_LE as a valid format option. Results with skylake-driver and snd_hda_intel show status quo on our RVPs. PR filed on SOF github shows promising results too [4]. Changes in v2: - patch 01/17, introduced struct snd_pcm_subformat which task is to represent subformat-mask on per format basis. Expectation is that manipulated arrays of subformats always end with a sentinel entry - patch 01/17, added snd_pcm_hw_copy() as the copying snd_pcm_hardware becomes non-trivial - patch 02/17, added hw_rule that produces final subformat mask based on provided formats as suggested by Jaroslav - patch 04/17, soc_pcm_hw_update_subformat() refactored as the subformat intersection becomes non-trivial - relevant functions releasing resources occupied by hda_pcm and snd_pcm_runtime updated to also kfree() subformats - except for 16/17, no changes to patches past 04/17, retaining acks for these Changes in v1: - fixed UBSAN due to missing snd_pcm_subformat_names[] entries for new subformats - as HDMI stream capabilities are assigned on PCM open, patch 16/17 has been updated to ignore such codecs for now. A separate patchset will take care of this case - params_bps() reworded to snd_pcm_hw_params_bps() - fixed compilation issues in sof-driver, patch 13/17 [1]: https://lore.kernel.org/alsa-devel/20190905053302.9262-1-pawel.harlozinski@… [2]: https://github.com/alsa-project/alsa-utils/pull/228 [3]: https://github.com/alsa-project/alsa-lib/pull/342 [4]: https://github.com/thesofproject/linux/pull/4539 Cezary Rojewski (17): ALSA: pcm: Introduce MSBITS subformat interface ALSA: pcm: Honor subformat when configuring substream ALSA: hda: Honor subformat when querying PCMs ASoC: pcm: Honor subformat when configuring runtime ALSA: hda: Upgrade stream-format infrastructure ALSA: hda: Switch to new stream-format interface ALSA: hda/hdmi: Switch to new stream-format interface ALSA: hda/ca0132: Switch to new stream-format interface ASoC: codecs: hda: Switch to new stream-format interface ASoC: codecs: hdac_hda: Switch to new stream-format interface ASoC: codecs: hdac_hdmi: Switch to new stream-format interface ASoC: Intel Skylake: Switch to new stream-format interface ASoC: SOF: Intel: Switch to new stream-format interface ASoC: Intel: avs: Switch to new stream-format interface ALSA: hda: Drop snd_hdac_calc_stream_format() ASoC: Intel: avs: Kill S24_LE in HDAudio streaming ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description include/sound/hda_codec.h | 5 +- include/sound/hdaudio.h | 14 +-- include/sound/pcm.h | 14 +++ include/sound/pcm_params.h | 2 + include/sound/soc.h | 1 + include/uapi/sound/asound.h | 7 +- sound/core/pcm.c | 4 + sound/core/pcm_lib.c | 28 +++++ sound/core/pcm_misc.c | 51 ++++++++++ sound/core/pcm_native.c | 67 +++++++++++- sound/hda/hdac_device.c | 163 ++++++++++++++++++++++-------- sound/pci/hda/hda_codec.c | 5 + sound/pci/hda/hda_controller.c | 10 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_hdmi.c | 6 +- sound/soc/codecs/hda-dai.c | 6 +- sound/soc/codecs/hda.c | 2 + sound/soc/codecs/hdac_hda.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 10 +- sound/soc/intel/avs/loader.c | 4 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 54 ++++++++-- sound/soc/intel/avs/probes.c | 3 +- sound/soc/intel/avs/topology.c | 19 +++- sound/soc/intel/skylake/skl-pcm.c | 11 +- sound/soc/soc-pcm.c | 35 ++++++- sound/soc/sof/intel/hda-dai-ops.c | 5 +- tools/include/uapi/sound/asound.h | 7 +- 28 files changed, 443 insertions(+), 103 deletions(-) base-commit: 564ee9ac788b680e4ec4a6cb3a4a953dc61d5da8 -- 2.25.1

4 25

[PATCH v3 00/42] ep93xx device tree conversion
by Nikita Shubin via B4 Relay 13 Nov '23

13 Nov '23

This series aims to convert ep93xx from platform to full device tree support. The main goal is to receive ACK's to take it via Arnd's arm-soc branch. I've moved to b4, tricking it to consider v0 as v1, so it consider's this version to be v3, this exactly the third version. Link to previous versions: https://lore.kernel.org/all/20230424123522.18302-1-nikita.shubin@maquefel.m… Major changes with clocks: - split all ep93xx clk into two groups: - those who are special to ep93xx (i.e. can't be implemented via standard clk's) and use special swlocked registers - those who can be implemented via devm_clk_hw_register_gate, etc... - moved all clk's that are not used via device tree to SoC driver - dropped init and globals - devm_* everywhere Other notable changes: - added a global spinlock for SoC driver to protect swlocked register's. - moved dts file into cirrus - dropped "dt-bindings: gpio: Add Cirrus EP93xx" as it was taken by Bartosz via gpio subsystem on master v6.5-rc2-89-g3acbd4602b05: make dtbs_check ``` make -j34 -C build-linux ARCH=arm CROSS_COMPILE=armv4t-softfloat-linux-gnueabi- dtbs_check -s arch/arm/boot/dts/cirrus/ep93xx-edb9302.dtb:0:0: /soc/spi@808a0000/codec@0: failed to match any schema with compatible: ['cirrus,cs4271'] /home/maquefel/workshop/ts7250-boot-build/build-linux/arch/arm/boot/dts/cirrus/ep93xx-edb9302.dtb: sound: 'simple-audio-card,convert-sample-format' does not match any of the regexes: '^simple-audio-card,codec(@[0-9a-f]+)?$', '^simple-audio-card,cpu(@[0-9a-f]+)?$', '^simple-audio-card,dai-link(@[0-9a-f]+)?$', '^simple-audio-card,plat(@[0-9a-f]+)?$', 'pinctrl-[0-9]+' From schema: /home/maquefel/workshop/ts7250-boot-build/linux/Documentation/devicetree/bindings/sound/simple-card.yaml ``` cirrus,cs4271 is defined in linux/Documentation/devicetree/bindings/sound/cs4271.txt `make dt_binding_check` shows no issues. [PATCH v1 01/43] gpio: ep93xx: split device in multiple - core.c: - fixed tab used instead of space - unsigned i in ep93xx_init_devices - gpio-ep93xx.c: - single line for EP93XX_INT_DEBOUNCE_OFFSET - drop assignment from definition - s/devm_kcalloc/devm_kzalloc/ - value check for platform_get_irq - used dev_err_probe - s/if (irq <= 0)/if (irq < 0)/ [PATCH v1 02/43] dt-bindings: soc: Add Cirrus EP93xx - fixed title - dropped "+|" - dropped example, syscon, timer, required - dropped clock header from this patch and moved to clk patch - don't shadow irq_of_parse_and_map error NOTES: There is no possiblity to use fwnode_irq_get for TIMER_OF_DECLARE. [PATCH v1 03/43] soc: Add SoC driver for Cirrus ep93xx - s/Soc/SoC/ - sorted Kconfig, Makefile - s/linux/Linux/ - headers sorted - drop parentness - drop comment entirely (it's obvious) - drop global map - add SoC Bus stuff to get rid of exporting ep93xx_chip_revision Only PATA and CLK currently rely on revision, they can take it from soc_device_match. [PATCH v1 04/43] dt-bindings: clock: Add Cirrus EP93xx - filename now matches compatible - dropped description - dropped clock description - dropped fixed-clock from example - compatible first in example [PATCH v1 06/43] dt-bindings: pinctrl: Add Cirrus EP93xx - blank line after function - dropped minItems, maxItems for groups - fixed weird wrapping [PATCH v1 07/43] pinctrl: add a Cirrus ep93xx SoC pin controller Nikita Shubin - dropped 0 assigment in enum ep93xx_pinctrl_model - convert to pingroup - convert to pinfunction - str_enabled_disabled - s/of_match_node/of_device_get_match_data/ - numerous defaults for cases - convert to dev_err_probe - dropped dev_info noise - add "gpio2agrp" to ep9301/ep9302 as they have 0 pin of Port C exposed [PATCH v1 09/43] clocksource: ep93xx: Add driver for Cirrus Logic EP93xx - headers sorted - fixed comment - use lo_hi_readq and UINT_MAX [PATCH 12/43] clk: ep93xx: add DT support for Cirrus EP93xx - move out all clocks that aren't used in device tree to SoC driver (pll1, pll2, usb) - drop all static variables (regmap, clock data) - DMA and USB clocks as standart gates to avoid additional checks (all other require swlocked_write) - rearranged ep93xx_clk members to avoid holes - numerous fixes from Andy - don't get revision, use soc_device_match instead - added own clk_get function to distinguish fixed and own clocks [PATCH v1 14/43] power: reset: Add a driver for the ep93xx reset - added private data with regmap, as we dropped global regmap in SoC dirver - use dev_err_probe - + bits.h header [PATCH v1 17/43] dt-bindings: spi: Add Cirrus EP93xx - fixed filename to cirrus,ep9301-spi.yaml - dropped address and size cells - dropped cs-gpio - dropped cirrus,ep9301-use-dma - we are looking into "dmas" property now - added proper gpio defines [PATCH v1 18/43] spi: ep93xx: add DT support for Cirrus EP93xx - use device_set_node + dev_fwnode - s/of_property_present/device_property_present/ [PATCH v1 20/43] net: cirrus: add DT support for Cirrus EP93xx - +dev_err_probe [PATCH v1 22/43] dma: cirrus: add DT support for Cirrus EP93xx - dropped subsys_initcall probe - use of_device_get_match_data - use struct_size for allocating memory with channel array - rearranged data to take it out of CONFIG_OF - + dev_err_probe [PATCH v1 23/43] dt-bindings: mtd: Add ts7250 nand-controller - unevaluatedProperties: false - dropped address and size cells - added nand node [PATCH v1 24/43] mtd: nand: add support for ts72xx - return -EINVAL on engine_type == ON_HOST - added err_nand_cleanup goto - s/derived/Derived from/ - BIT and GENMASK - s/sizeof(struct ts72xx_nand_data)/sizeof(*data)/ and one line - fixed wierd mtd_device_parse_register error handling - added child node parsing to conform device tree bindings (i.e. partitions contained in 'nand@0' child node) [PATCH v1 26/43] pata: cirrus: add DT support for Cirrus EP93xx - headers reordering - s/-ENXIO/-ENOMEM/ - drop ep93xx_chip_revision, get data from soc_device_match - changed title to "ata: pata_ep93xx: add device tree support" [PATCH v1 27/43] dt-bindings: input: Add Cirrus EP93xx keypad - s/cirrus,debounce-delay-ms/debounce-delay-ms/ - cirrus,prescale -> uint16 and constraints - added "autorepeat" - fixed compatible in example [PATCH v1 28/43] input: keypad: ep93xx: add DT support for Cirrus EP93xx - added autorepeat - dropped module param flags entirely - s/of_property_read/device_property_read/ [PATCH v1 29/43] dt-bindings: rtc: Add ST M48T86 - renamed file to st,m48t86.yaml - fixed typo - reordered props [PATCH v1 33/43] dt-bindings: gpio: Add Cirrus EP93xx - renamed file to cirrus,ep9301-gpio.yaml - dropped explicit examples (but left port F with uses 8 interrupts) [PATCH v1 34/43] gpio: ep93xx: add DT support for gpio-ep93xx - s/Add match table./Add OF ID match table./ [PATCH v1 35/43] ARM: dts: add device tree for ep93xx Soc - splitted dtsi and dts - made nand-controller, leds, etc... part of the board and not SoC - dropped empty spi bus - labels sorted - dropped "delete-property" everywhere - compatible first, reg is second, ranges, if present, third - fixed wrapping - status last - moved cs gpios to board files - led0 -> led-0, led1 -> led-1 [PATCH v1 38/43] pata: cirrus: drop legacy pinctrl - changed title to "ata: pata_ep93xx: remove legacy pinctrl use" - removed ep93xx_ide_acquire_gpio and ep93xx_ide_release_gpio along with this patch -- 2.39.2 --- Alexander Sverdlin (2): ARM: dts: ep93xx: Add EDB9302 DT ASoC: cirrus: edb93xx: Delete driver Nikita Shubin (40): gpio: ep93xx: split device in multiple dt-bindings: clock: Add Cirrus EP93xx clk: ep93xx: add DT support for Cirrus EP93xx dt-bindings: pinctrl: Add Cirrus EP93xx pinctrl: add a Cirrus ep93xx SoC pin controller dt-bindings: soc: Add Cirrus EP93xx soc: Add SoC driver for Cirrus ep93xx dt-bindings: timers: Add Cirrus EP93xx clocksource: ep93xx: Add driver for Cirrus Logic EP93xx dt-bindings: rtc: Add Cirrus EP93xx rtc: ep93xx: add DT support for Cirrus EP93xx dt-bindings: watchdog: Add Cirrus EP93x watchdog: ep93xx: add DT support for Cirrus EP93xx power: reset: Add a driver for the ep93xx reset dt-bindings: pwm: Add Cirrus EP93xx pwm: ep93xx: add DT support for Cirrus EP93xx dt-bindings: spi: Add Cirrus EP93xx spi: ep93xx: add DT support for Cirrus EP93xx dt-bindings: net: Add Cirrus EP93xx net: cirrus: add DT support for Cirrus EP93xx dt-bindings: dma: Add Cirrus EP93xx dma: cirrus: add DT support for Cirrus EP93xx dt-bindings: mtd: Add ts7200 nand-controller mtd: nand: add support for ts72xx dt-bindings: ata: Add Cirrus EP93xx ata: pata_ep93xx: add device tree support dt-bindings: input: Add Cirrus EP93xx keypad input: keypad: ep93xx: add DT support for Cirrus EP93xx dt-bindings: rtc: Add ST M48T86 rtc: m48t86: add DT support for m48t86 dt-bindings: wdt: Add ts72xx wdt: ts72xx: add DT support for ts72xx gpio: ep93xx: add DT support for gpio-ep93xx ARM: dts: add Cirrus EP93XX SoC .dtsi ARM: dts: ep93xx: add ts7250 board ARM: ep93xx: DT for the Cirrus ep93xx SoC platforms pwm: ep93xx: drop legacy pinctrl ata: pata_ep93xx: remove legacy pinctrl use ARM: ep93xx: delete all boardfiles ARM: ep93xx: soc: drop defines .../bindings/arm/cirrus/ep9301-syscon.yaml | 59 + .../devicetree/bindings/arm/cirrus/ep9301.yaml | 39 + .../bindings/ata/cirrus,ep9312-pata.yaml | 44 + .../bindings/clock/cirrus,ep9301-clk.yaml | 46 + .../bindings/dma/cirrus,ep9301-dma-m2m.yaml | 72 + .../bindings/dma/cirrus,ep9301-dma-m2p.yaml | 124 ++ .../bindings/input/cirrus,ep9307-keypad.yaml | 87 ++ .../devicetree/bindings/mtd/technologic,nand.yaml | 47 + .../devicetree/bindings/net/cirrus,ep9301-eth.yaml | 61 + .../bindings/pinctrl/cirrus,ep9301-pinctrl.yaml | 58 + .../devicetree/bindings/pwm/cirrus,ep9301-pwm.yaml | 48 + .../devicetree/bindings/rtc/cirrus,ep9301-rtc.yaml | 40 + .../devicetree/bindings/rtc/st,m48t86.yaml | 38 + .../devicetree/bindings/spi/cirrus,ep9301-spi.yaml | 63 + .../bindings/timer/cirrus,ep9301-timer.yaml | 49 + .../bindings/watchdog/cirrus,ep9301-wdt.yaml | 46 + .../bindings/watchdog/technologic,ts7200-wdt.yaml | 46 + arch/arm/Makefile | 1 - arch/arm/boot/dts/cirrus/Makefile | 4 + arch/arm/boot/dts/cirrus/ep93xx-bk3.dts | 126 ++ arch/arm/boot/dts/cirrus/ep93xx-edb9302.dts | 178 +++ arch/arm/boot/dts/cirrus/ep93xx-ts7250.dts | 145 ++ arch/arm/boot/dts/cirrus/ep93xx.dtsi | 449 ++++++ arch/arm/mach-ep93xx/Kconfig | 20 +- arch/arm/mach-ep93xx/Makefile | 11 - arch/arm/mach-ep93xx/clock.c | 733 ---------- arch/arm/mach-ep93xx/core.c | 1017 -------------- arch/arm/mach-ep93xx/dma.c | 114 -- arch/arm/mach-ep93xx/edb93xx.c | 344 ----- arch/arm/mach-ep93xx/ep93xx-regs.h | 38 - arch/arm/mach-ep93xx/gpio-ep93xx.h | 111 -- arch/arm/mach-ep93xx/hardware.h | 25 - arch/arm/mach-ep93xx/irqs.h | 76 - arch/arm/mach-ep93xx/platform.h | 42 - arch/arm/mach-ep93xx/soc.h | 212 --- arch/arm/mach-ep93xx/ts72xx.c | 422 ------ arch/arm/mach-ep93xx/ts72xx.h | 94 -- arch/arm/mach-ep93xx/vision_ep9307.c | 311 ----- drivers/ata/pata_ep93xx.c | 49 +- drivers/clk/Kconfig | 8 + drivers/clk/Makefile | 1 + drivers/clk/clk-ep93xx.c | 764 +++++++++++ drivers/clocksource/Kconfig | 11 + drivers/clocksource/Makefile | 1 + .../clocksource}/timer-ep93xx.c | 155 ++- drivers/dma/ep93xx_dma.c | 131 +- drivers/gpio/gpio-ep93xx.c | 332 ++--- drivers/input/keyboard/ep93xx_keypad.c | 74 +- drivers/mtd/nand/raw/Kconfig | 7 + drivers/mtd/nand/raw/Makefile | 1 + drivers/mtd/nand/raw/technologic-nand-controller.c | 162 +++ drivers/net/ethernet/cirrus/ep93xx_eth.c | 61 +- drivers/pinctrl/Kconfig | 7 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-ep93xx.c | 1447 ++++++++++++++++++++ drivers/power/reset/Kconfig | 10 + drivers/power/reset/Makefile | 1 + drivers/power/reset/ep93xx-restart.c | 86 ++ drivers/pwm/pwm-ep93xx.c | 26 +- drivers/rtc/rtc-ep93xx.c | 8 + drivers/rtc/rtc-m48t86.c | 8 + drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + drivers/soc/cirrus/Kconfig | 12 + drivers/soc/cirrus/Makefile | 2 + drivers/soc/cirrus/soc-ep93xx.c | 231 ++++ drivers/spi/spi-ep93xx.c | 28 +- drivers/watchdog/ep93xx_wdt.c | 8 + drivers/watchdog/ts72xx_wdt.c | 8 + include/dt-bindings/clock/cirrus,ep93xx-clock.h | 41 + include/dt-bindings/dma/cirrus,ep93xx-dma.h | 26 + include/linux/platform_data/dma-ep93xx.h | 25 +- include/linux/platform_data/eth-ep93xx.h | 10 - include/linux/platform_data/keypad-ep93xx.h | 32 - include/linux/soc/cirrus/ep93xx.h | 33 +- sound/soc/cirrus/Kconfig | 9 - sound/soc/cirrus/Makefile | 4 - sound/soc/cirrus/edb93xx.c | 117 -- 78 files changed, 5205 insertions(+), 4174 deletions(-) --- base-commit: ccff6d117d8dc8d8d86e8695a75e5f8b01e573bf change-id: 20230605-ep93xx-01c76317e2d2 Best regards, -- Nikita Shubin <nikita.shubin(a)maquefel.me>

15 90

[PATCH v2 1/2] ALSA: docs: Add Marian M2 driver documentation
by Ivan Orlov 05 Nov '23

05 Nov '23

Add documentation for the new MARIAN Seraph M2 sound card. It covers current development status, available controls of the card and information about the integrated loopback. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Remove redundant documentation fix from the next patch in the series Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/marian-m2.rst | 104 ++++++++++++++++++++++++ 2 files changed, 105 insertions(+) create mode 100644 Documentation/sound/cards/marian-m2.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e68bbb13c384..e873592d8d00 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -19,3 +19,4 @@ Card-Specific Information serial-u16550 img-spdif-in pcmtest + marian-m2 diff --git a/Documentation/sound/cards/marian-m2.rst b/Documentation/sound/cards/marian-m2.rst new file mode 100644 index 000000000000..bf12445e20d7 --- /dev/null +++ b/Documentation/sound/cards/marian-m2.rst @@ -0,0 +1,104 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +MARIAN Seraph M2 Driver +======================= + +Sep 18, 2023 + +Ivan Orlov <ivan.orlov0322(a)gmail.com> + +STATE OF DEVELOPMENT +==================== + +This driver is based on the driver written by Florian Faber in 2012, which seemed to work fine. +However, the initial code contained multiple issues, which had to be solved before sending the +driver upstream. + +The vendor lost the full documentation, so what we have here was recovered from drafts and found +after experiments with the card. + +What seems to be working fine: +- Playback and capture for all supported rates +- Integrated loopback (with some exceptions, see below) + +MEMORY MODEL +============ + +The hardware requires one huge contiguous DMA space to be allocated. After allocation, the bus address of +this buffer should be written to the hardware register. + +We can split this space into two parts: the first one contains samples for capture, another one contains +play samples: + +CAPTURE_CH_0, CAPTURE_CH_1, ..., CAPTURE_CH_127 | PLAY_CH_0, PLAY_CH_1, ..., PLAY_CH_127 + +The card supports the non-interleaved access mode only, so samples for each channel lay together: + +C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 | C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 + +The count of samples per each channel buffer needs to be set explicitly, so the address of the first +byte of the playback data depends on this value. The playback buffer starts where the capture buffer ends. +It makes the arbitrary period count/buffer size feature impossible to implement, and the driver supports only +2 periods per buffer. + +Controls +======== + +Input 1 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal +Input 2 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal + +Input 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Input 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Input 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Input 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Input 1 Frequency + Measured frequency on Input 1 +Input 2 Frequency + Measured frequency on Input 2 + +Output 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Output 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Output 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Output 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Clock Source + Internal/Sync Bus, Input Port 1, Input Port 2 + +Speed mode + VCO clock range (slow/fast) + +DCO Frequency (Hz) + +DCO Frequency (ms) + +DCO Detune + Range: -200...200 + +Loopback + Enable/Disable integrated loopback + + +Loopback +======== + +The card contains integrated loopback. When it is enabled, it sets the hardware’s DAW-in memory pointer +to the hardware’s DAW-out memory. So, what you play is what you record. + +You can enable the integrated loopback using the corresponding control. + +The loopback seems to work well on lower rates (like 28000). However, when the rate goes higher, I observe +multiple mistakes in recorded byte ordering. -- 2.34.1

5 12

Re: RFC: usb: gadget: u_audio: Notifying gadget that host started playback/capture?
by Pavel Hofman 30 Oct '23

30 Oct '23

Hi, Dne 08. 09. 21 v 10:21 Pavel Hofman napsal(a): > Hi, > > The current audio gadget has no way to inform the gadget side that the > host side has started playback/capture and that gadget-side alsa > processes should be started. > > Playback/capture processes on the host side do not get stuck without the > gadget side consuming/producing data (OUT requests are ignored in > u_audio_iso_complete, IN ones send initial zeros in their req->buf). > > However, playback/capture processes on the gadget side get stuck without > the host side sending playback OUT packets or capture IN requests and > time out with error. If there was a way to inform the gadget side that > playback/capture has started on the host side, the gadget clients could > react accordingly. > I drafted a simple patch for u_audio.c which defines read-only boolean ctl elems "Capture Requested" and "Playback Requested". Their values are set/reset in methods u_audio_start_capture/playback and u_audio_stop_capture/playback, i.e. at changes of respective altsettings from 0 to 1 and back. Every ctl elem value change sends notification via snd_ctl_notify. The principle works OK for capture/playback start/stop on the host, as monitored by alsactl: pi@raspberrypi:~ $ alsactl monitor hw:UAC2Gadget node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE However at enumeration the USB host switches both playback and capture altsettings repeatedly, generating "fake" events from the gadget side POW. The host even sends regular-sized EP-OUT packets filled with zeros during enumeration (tested on linux only for now). Please is there any way to "detect" the enumeration stage to mask out the "fake" playback/capture start/stop events? The attached patch does not apply cleanly to mainline u_audio.c because it's rebased on other patches not submitted yet but it's only a discussion inducer for now. Thanks a lot, Pavel.

2 6

[PATCH] patch_realtek: Splitting the UX3402 into two separate models.
by poseaydone＠ya.ru 28 Oct '23

28 Oct '23

From: Knyazev Arseniy <poseaydone(a)ya.ru> UX3402VA and UX3402ZA models require different hex values, so comibining them into one model is incorrect. Signed-off-by: Knyazev Arseniy <poseaydone(a)ya.ru> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b7e78bfcffd8..b002f947b0d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9812,7 +9812,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS ROG Strix G17 2023 (G713PV)", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), - SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS UX3402VA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), -- 2.42.0

3 2

[PATCH 0/4] ASoC: qcom: fixes for Click/Pop Noise
by Srinivas Kandagatla 27 Oct '23

27 Oct '23

Click/Pop Noise was a long pending issue with WSA Codecs which are prone to accumlate DC when ports are active but without any data streams. There are multiple places in the current setup, where this could happen in both startup as well as shutdown path. This patches help fix those issues by making sure the PA is Muted/Unmuted inline with the stream start/stop events. Other than this there is a patch to fix DSP graph saturation issue as we never unload playback graph before loading same graph. Srinivas Kandagatla (4): ASoC: qcom: q6apm-lpass-dai: close graphs before opening a new one ASoC: qcom: sdw: do not restart soundwire ports for every prepare ASoC: codecs: wsa883x: mute/unmute PA in correct sequence ASoC: codecs: wsa881x: mute/unmute PA in correct sequence sound/soc/codecs/wsa881x.c | 30 ++++++++++++++++++++- sound/soc/codecs/wsa883x.c | 36 ++++++++++++++++++++----- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 3 +++ sound/soc/qcom/sdw.c | 7 ++--- 4 files changed, 63 insertions(+), 13 deletions(-) -- 2.21.0

4 16

[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

3 2

[PATCH 0/3] ARM: omap4: embt2ws: Add audio support
by Andreas Kemnade 25 Oct '23

25 Oct '23

Add audio support for Epson Moverio BT-200. In the vendor kernel, the MCBSP side is used as master, so do it here also that way. Andreas Kemnade (3): ASoC: ti: omap-mcbsp: Ignore errors for getting fck_src ASoC: tlv320aic3x: use BCLK instead of MCLK if not in master mode ARM: dts: omap4: embt2ws: Add audio support arch/arm/boot/dts/omap4-epson-embt2ws.dts | 21 +++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/ti/omap-mcbsp.c | 4 ++-- 3 files changed, 27 insertions(+), 2 deletions(-) -- 2.39.2

4 28

[PATCH v4 00/42] ep93xx device tree conversion
by Nikita Shubin via B4 Relay 16 Oct '23

16 Oct '23

This series aims to convert ep93xx from platform to full device tree support. The main goal is to receive ACK's to take it via Arnd's arm-soc branch. Major changes: - drop newline at the end from each YAML files - rename dma and clk bindings headers to match first compatible - shrink SoC exported functions number to only 2 - dropped some ep93xx_pata fixes from these series - dropped m48t86 stuff from these series Bit thanks to Andy Shevchenko for thorough review. Some parts are were sent in advance, so these series depends on the following: Link: https://lore.kernel.org/all/20230823-m48t86_device_tree-v2-0-21ff275f949d@m… Link: https://lore.kernel.org/all/20230823-ep93xx_pata_fixes-v1-0-d7e7229be148@ma… --- Changes in v4: - gpio: ep93xx: split device in multiple - s/generic_handle_irq/generic_handle_domain_irq/ - s/int offset/irq_hw_number_t offset/ though now it looks a bit odd to me - drop i = 0 - drop 'error' - use dev_err_probe withour printing devname once again dt-bindings: clock: Add Cirrus EP93xx - renamed cirrus,ep93xx-clock.h -> cirrus,ep9301-clk.h clk: ep93xx: add DT support for Cirrus EP93xx - drop unused includes - use .name only for xtali, pll1, pll2 parents - convert // to /* - pass clk_parent_data instead of char* clock name dt-bindings: pinctrl: Add Cirrus EP93xx - s/additionalProperties/unevaluatedProperties/ dt-bindings: soc: Add Cirrus EP93xx - move syscon to soc directory - add vendor prefix - make reboot same style as pinctrl, clk - use absolute path for ref - expand example soc: Add SoC driver for Cirrus ep93xx - s/0xf0000000/GENMASK(31, 28)/ - s/ret/ep93xx_chip_revision(map)/ - drop symbol exports - convert to platform driver dt-bindings: rtc: Add Cirrus EP93xx - allOf: with $ref to rtc.yaml - s/additionalProperties/unevaluatedProperties/ dt-bindings: watchdog: Add Cirrus EP93x - drop description - reword power: reset: Add a driver for the ep93xx reset - lets use 'GPL-2.0+' instead of '(GPL-2.0)' - s/of_device/of/ - drop mdelay with warning - return 0 at the end net: cirrus: add DT support for Cirrus EP93xx - fix leaking np mtd: nand: add support for ts72xx - +bits.h - drop comment - ok to fwnode_get_next_child_node - use goto to put handle and nand and report error ARM: dts: add Cirrus EP93XX SoC .dtsi - add simple-bus for ebi, as we don't require to setup anything - add arm,pl011 compatible to uart nodes - drop i2c-gpio, as it's isn't used anywhere ARM: dts: ep93xx: add ts7250 board - generic node name for temperature-sensor - drop i2c - move nand, rtc, watchdog to ebi node - Link to v3: https://lore.kernel.org/r/20230605-ep93xx-v3-0-3d63a5f1103e@maquefel.me --- Alexander Sverdlin (2): ARM: dts: ep93xx: Add EDB9302 DT ASoC: cirrus: edb93xx: Delete driver Nikita Shubin (40): gpio: ep93xx: split device in multiple ARM: ep93xx: add swlocked prototypes dt-bindings: clock: Add Cirrus EP93xx clk: ep93xx: add DT support for Cirrus EP93xx dt-bindings: pinctrl: Add Cirrus EP93xx pinctrl: add a Cirrus ep93xx SoC pin controller dt-bindings: power: reset: Add ep93xx reset power: reset: Add a driver for the ep93xx reset dt-bindings: soc: Add Cirrus EP93xx soc: Add SoC driver for Cirrus ep93xx dt-bindings: timers: Add Cirrus EP93xx clocksource: ep93xx: Add driver for Cirrus Logic EP93xx dt-bindings: rtc: Add Cirrus EP93xx rtc: ep93xx: add DT support for Cirrus EP93xx dt-bindings: watchdog: Add Cirrus EP93x watchdog: ep93xx: add DT support for Cirrus EP93xx dt-bindings: pwm: Add Cirrus EP93xx pwm: ep93xx: add DT support for Cirrus EP93xx dt-bindings: spi: Add Cirrus EP93xx spi: ep93xx: add DT support for Cirrus EP93xx dt-bindings: net: Add Cirrus EP93xx net: cirrus: add DT support for Cirrus EP93xx dt-bindings: dma: Add Cirrus EP93xx dma: cirrus: add DT support for Cirrus EP93xx dt-bindings: mtd: Add ts7200 nand-controller mtd: nand: add support for ts72xx dt-bindings: ata: Add Cirrus EP93xx ata: pata_ep93xx: add device tree support dt-bindings: input: Add Cirrus EP93xx keypad input: keypad: ep93xx: add DT support for Cirrus EP93xx dt-bindings: wdt: Add ts72xx wdt: ts72xx: add DT support for ts72xx gpio: ep93xx: add DT support for gpio-ep93xx ARM: dts: add Cirrus EP93XX SoC .dtsi ARM: dts: ep93xx: add ts7250 board ARM: ep93xx: DT for the Cirrus ep93xx SoC platforms pwm: ep93xx: drop legacy pinctrl ata: pata_ep93xx: remove legacy pinctrl use ARM: ep93xx: delete all boardfiles ARM: ep93xx: soc: drop defines .../bindings/arm/cirrus/cirrus,ep9301.yaml | 38 + .../bindings/ata/cirrus,ep9312-pata.yaml | 42 + .../bindings/clock/cirrus,ep9301-clk.yaml | 46 + .../bindings/dma/cirrus,ep9301-dma-m2m.yaml | 69 + .../bindings/dma/cirrus,ep9301-dma-m2p.yaml | 121 ++ .../bindings/input/cirrus,ep9307-keypad.yaml | 87 ++ .../devicetree/bindings/mtd/technologic,nand.yaml | 45 + .../devicetree/bindings/net/cirrus,ep9301-eth.yaml | 59 + .../bindings/pinctrl/cirrus,ep9301-pinctrl.yaml | 57 + .../bindings/power/reset/cirrus,ep9301-reboot.yaml | 34 + .../devicetree/bindings/pwm/cirrus,ep9301-pwm.yaml | 46 + .../devicetree/bindings/rtc/cirrus,ep9301-rtc.yaml | 42 + .../bindings/soc/cirrus/cirrus,ep9301-syscon.yaml | 71 + .../devicetree/bindings/spi/cirrus,ep9301-spi.yaml | 61 + .../bindings/timer/cirrus,ep9301-timer.yaml | 49 + .../bindings/watchdog/cirrus,ep9301-wdt.yaml | 42 + .../bindings/watchdog/technologic,ts7200-wdt.yaml | 45 + arch/arm/Makefile | 1 - arch/arm/boot/dts/cirrus/Makefile | 4 + arch/arm/boot/dts/cirrus/ep93xx-bk3.dts | 124 ++ arch/arm/boot/dts/cirrus/ep93xx-edb9302.dts | 180 +++ arch/arm/boot/dts/cirrus/ep93xx-ts7250.dts | 143 ++ arch/arm/boot/dts/cirrus/ep93xx.dtsi | 454 +++++++ arch/arm/mach-ep93xx/Kconfig | 20 +- arch/arm/mach-ep93xx/Makefile | 11 - arch/arm/mach-ep93xx/clock.c | 733 ---------- arch/arm/mach-ep93xx/core.c | 1017 -------------- arch/arm/mach-ep93xx/dma.c | 114 -- arch/arm/mach-ep93xx/edb93xx.c | 344 ----- arch/arm/mach-ep93xx/ep93xx-regs.h | 38 - arch/arm/mach-ep93xx/gpio-ep93xx.h | 111 -- arch/arm/mach-ep93xx/hardware.h | 25 - arch/arm/mach-ep93xx/irqs.h | 76 -- arch/arm/mach-ep93xx/platform.h | 42 - arch/arm/mach-ep93xx/soc.h | 212 --- arch/arm/mach-ep93xx/ts72xx.c | 422 ------ arch/arm/mach-ep93xx/ts72xx.h | 94 -- arch/arm/mach-ep93xx/vision_ep9307.c | 311 ----- drivers/ata/pata_ep93xx.c | 33 +- drivers/clk/Kconfig | 8 + drivers/clk/Makefile | 1 + drivers/clk/clk-ep93xx.c | 753 +++++++++++ drivers/clocksource/Kconfig | 11 + drivers/clocksource/Makefile | 1 + .../clocksource}/timer-ep93xx.c | 155 ++- drivers/dma/ep93xx_dma.c | 125 +- drivers/gpio/gpio-ep93xx.c | 331 ++--- drivers/input/keyboard/ep93xx_keypad.c | 74 +- drivers/mtd/nand/raw/Kconfig | 7 + drivers/mtd/nand/raw/Makefile | 1 + drivers/mtd/nand/raw/technologic-nand-controller.c | 166 +++ drivers/net/ethernet/cirrus/ep93xx_eth.c | 63 +- drivers/pinctrl/Kconfig | 7 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-ep93xx.c | 1429 ++++++++++++++++++++ drivers/power/reset/Kconfig | 10 + drivers/power/reset/Makefile | 1 + drivers/power/reset/ep93xx-restart.c | 85 ++ drivers/pwm/pwm-ep93xx.c | 26 +- drivers/rtc/rtc-ep93xx.c | 8 + drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + drivers/soc/cirrus/Kconfig | 12 + drivers/soc/cirrus/Makefile | 2 + drivers/soc/cirrus/soc-ep93xx.c | 215 +++ drivers/spi/spi-ep93xx.c | 28 +- drivers/watchdog/ep93xx_wdt.c | 8 + drivers/watchdog/ts72xx_wdt.c | 8 + include/dt-bindings/clock/cirrus,ep9301-clk.h | 41 + include/dt-bindings/dma/cirrus,ep93xx-dma.h | 26 + include/linux/platform_data/dma-ep93xx.h | 25 +- include/linux/platform_data/eth-ep93xx.h | 10 - include/linux/platform_data/keypad-ep93xx.h | 32 - include/linux/soc/cirrus/ep93xx.h | 29 +- sound/soc/cirrus/Kconfig | 9 - sound/soc/cirrus/Makefile | 4 - sound/soc/cirrus/edb93xx.c | 117 -- 77 files changed, 5126 insertions(+), 4168 deletions(-) --- base-commit: bdc09c8e8b16d494ccd1c56e903e78dd76455a35 change-id: 20230605-ep93xx-01c76317e2d2 Best regards, -- Nikita Shubin <nikita.shubin(a)maquefel.me>

4 4

[PATCH] ASoC: apple: mca: Annotate struct mca_data with __counted_by
by Kees Cook 11 Oct '23

11 Oct '23

Prepare for the coming implementation by GCC and Clang of the __counted_by attribute. Flexible array members annotated with __counted_by can have their accesses bounds-checked at run-time checking via CONFIG_UBSAN_BOUNDS (for array indexing) and CONFIG_FORTIFY_SOURCE (for strcpy/memcpy-family functions). As found with Coccinelle[1], add __counted_by for struct mca_data. [1] https://github.com/kees/kernel-tools/blob/trunk/coccinelle/examples/counted… Cc: "Martin Povišer" <povik+lin(a)cutebit.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: asahi(a)lists.linux.dev Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/soc/apple/mca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index ce77934f3eef..99e547ef95e6 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -161,7 +161,7 @@ struct mca_data { struct mutex port_mutex; int nclusters; - struct mca_cluster clusters[]; + struct mca_cluster clusters[] __counted_by(nclusters); }; static void mca_modify(struct mca_cluster *cl, int regoffset, u32 mask, u32 val) -- 2.34.1

3 7

[PATCH v7 00/30] Add support for QMC HDLC, framer infrastructure and PEF2256 framer
by Herve Codina 10 Oct '23

10 Oct '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4, 5, 6, 7) 2) Introduce the QMC HDLC driver (patches 8, 9, 10, 11) 3) Add QMC dynamic timeslot support (patches 12 - 22) 4) Add timeslots change support in QMC HDLC (patch 23) 5) Introduce framer infrastructure (patch 24) 6) Add PEF2256 framer provider (patches 25, 26, 27, 28) 7) Add framer codec as a framer consumer (patch 29) 8) Add framer support as a framer consumer in QMC HDLC (patch 30) The series contains the full story and detailed modifications. If needed, the series can be split and/or commits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230922075913.422435-1-herve.codina@b… This v7 series: - Fixes the framer binding removing the '$ref'. commit 033d0b1 ("Add '-bps' as a standard unit suffix for bits per second") was recently added in https://github.com/devicetree-org/dt-schema/ Even if not yet present in an dt-schema release, '-bps' is an official standard suffix and so '$ref' is no more needed. - Fixes the QMC channel binding. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v6 -> v7 - Patch 8 Move the 'compatible' property to the first property. Add device/SoC specific compatible strings. Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 25 Remove '$ref' from the lantiq,data-rate-bps property. '-bps' is now a standard suffix. Changes v5 -> v6 - All patches Remove 'Signed-off-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' These Sob were added by Christophe when he sent the v4 while I was on vacation. https://lore.kernel.org/linux-kernel/992a2b31-e21f-eee3-8bfc-a65b69fe5bd7@c… - Patch 6, 7 Add 'Acked-by: Conor Dooley <conor.dooley(a)microchip.com>' Add 'Acked-by: Rob Herring <robh(a)kernel.org>' - Patch 8 Add a constraint on fsl,framer. - Patch 25 Fix clocks description (inconsistent capitalisation) Use '8-bit' instead of '8bit' Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Removed patch 26 (patch applied) - Patch 26 (patch 27 in v5) Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 27 (patch 28 in v5) Move registers definition to .c file and remove .h file Add 'select PINCONF' in Kconfig Use dev_err_probe() Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 28 (patch 29 in v5) Remove .h file Changes v4 -> v5 - Patches 1 to 5 No changes - Patch 6 (new in v5) Fix QMC binding example - Patch 7 (new in v5) Add missing 'additionalProperties: false' - Patch 8 (new in v5, replace v4 patch 6) Add QMC HDLC properties in the QMC channel node Renamed the 'framer' property to 'fsl,framer' - Patch 9 (new in v5) Add support for QMC child devices - Patch 10 (patch 7 in v4) No changes - Patch 11 (patch 8 in v4) Remove fsl,qmc-hdlc.yaml (no more existing file) - Patches 12 to 22 (patches 9 to 19 in v4) No changes - Patch 23 (patch 20 in v4) Remove unused variable initializations Remove extra space - Patch 24 (patch 21 in v4) Improve Kconfig help text Fix variable declaration (reverse xmas tree) Fix typos and extra spaces Fix documentation issues raised by 'kernel-doc -none' Move of_node_put() and kfree() out of the mutex Replace ida_simple_{get,remove}() by ida_{alloc,free}() Support framer device-tree nodes without '#framer-cells' property - Patch 25 (patch 22 in v4) Fix $ref in the pinctrl subnode Remove '#framer-cells' property Add needed '|' - Patch 26 (patch 23 in v4) Fix a typo in the commit subject - Patches 27, 28, 29 (patch 24, 25, 26 in v4) No changes - Patch 30 (patch 27 in v4) Fix a typo in the commit log - Patch 31 (patch 28 in v4) Used 'fsl,framer' property name instead of 'framer' Changes v3 -> v4 - Patch 21 Fixes build failure with CONFIG_MODULES Changes v2 -> v3 - Patches 1, 2, 3, 4 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - New patch Remove inline keyword from the existing registers accessors helpers - Patch 6 (patches 5, 27 in v2) Update the binding title Squash patch 27 - Patch 7 (patch 6 in v2) Remove the cast in netdev_to_qmc_hdlc() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 8 (patch 7 in v2): No change - Patches 9, 10 (patches 8, 9 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 11 (patch 10 in v2) Remove inline keyword from the introduced qmc_clrsetbits16() helper Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patches 12, 13, 14, 15, 16, 17, 18, 19, 20 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 21 (patch 20 in v2) Remove unneeded framer NULL pointer check Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 22 (patch 21 in v2) Change sclkr and sclkx clocks description Remove the framer phandle property from the framer subnodes (ie. from framer-codec nodes) - Patch 23 (patch 22 in v2) Initialize 'disabled' variable at declaration Fix commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 24 (patch 23 in v2) Remove inline keyword from the existing registers accessors helpers Use dev_warn_ratelimited() in default interrupt handler Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 25 (patch 24 in v2) Replace #include "linux/bitfield.h" by #include <linux/bitfield.h> Fold the pinctrl anonymous struct into the struct pef2256_pinctrl Update commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 26 (patch 25 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 27 (patch 26 in v2) Fix error message Changed the ch.max computation in framer_dai_hw_rule_channels_by_format() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 28 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (30): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status soc: fsl: cpm1: qmc: Remove inline function specifiers dt-bindings: soc: fsl: cpm_qe: cpm1-scc-qmc: Fix example property name dt-bindings: soc: fsl: cpm_qe: cpm1-scc-qmc: Add 'additionalProperties: false' in child nodes dt-bindings: soc: fsl: cpm_qe: cpm1-scc-qmc: Add support for QMC HDLC soc: fsl: cpm1: qmc: Add support for child devices net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec net: wan: fsl_qmc_hdlc: Add framer support .../bindings/net/lantiq,pef2256.yaml | 213 +++++ .../soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml | 37 +- MAINTAINERS | 15 + drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 39 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 887 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 ++++++++++++++++ drivers/pinctrl/Kconfig | 15 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256.c | 358 +++++++ drivers/soc/fsl/qe/qmc.c | 592 +++++++++--- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 205 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 27 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 413 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 25 files changed, 4910 insertions(+), 140 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

4 39

[PATCH v2 0/4] Convert Rockchip ASoC drivers to GPIO descriptors
by Linus Walleij 09 Oct '23

09 Oct '23

The Rockchip drivers are pretty straight-forward to convert over to using GPIO descriptors. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Make the GPIOs properly optional in the first patch, preserving the old semantics of the driver. - Link to v1: https://lore.kernel.org/r/20230928-descriptors-asoc-rockchip-v1-0-a142a42d4… --- Linus Walleij (4): ASoC: rockchip: Convert RK3288 HDMI to GPIO descriptors ASoC: rockchip: Drop includes from RK3399 ASoC: rockchip: Drop includes from Rockchip MAX98090 ASoC: rockchip: Drop includes from Rockchip RT5645 sound/soc/rockchip/rk3288_hdmi_analog.c | 46 ++++++++++----------------------- sound/soc/rockchip/rk3399_gru_sound.c | 2 -- sound/soc/rockchip/rockchip_max98090.c | 2 -- sound/soc/rockchip/rockchip_rt5645.c | 2 -- 4 files changed, 14 insertions(+), 38 deletions(-) --- base-commit: 0bb80ecc33a8fb5a682236443c1e740d5c917d1d change-id: 20230926-descriptors-asoc-rockchip-833a4fbc7c1f Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 5

time to remove COPYING.GPL from alsa-plugins? (issue #45)
by Michael Wang 09 Oct '23

09 Oct '23

Hi Alsa-devs and Mr Iwai, Sorry for the spam, I am resending this in plain text mode this time. It seems the only thing in alsa-plugins that has a GPL license is rate/rate_samplerate.c, and that is due to the license of libsamplerate. But in 2016, libsamplerate dropped the commercial/GPL part of its license and was released with the 2-Clause BSD license. So I was wondering if the license for rate_samplerate.c can be updated to remove the GPL license, which means the COPYING.GPL file can also be removed from the alsa-plugins package? I have created issue #54 in github.com/alsa-project/alsa-plugins, but so far, no action. Should I submit a pull request? Thanks, Michael -- This electronic communication and the information and any files transmitted with it, or attached to it, are confidential and are intended solely for the use of the individual or entity to whom it is addressed and may contain information that is confidential, legally privileged, protected by privacy laws, or otherwise restricted from disclosure to anyone else. If you are not the intended recipient or the person responsible for delivering the e-mail to the intended recipient, you are hereby notified that any use, copying, distributing, dissemination, forwarding, printing, or copying of this e-mail is strictly prohibited. If you received this e-mail in error, please return the e-mail to the sender, delete it from your computer, and destroy any printed copy of it.

2 4

[PATCH v2 1/2] ASoC: dt-bindings: fsl_rpmsg: List DAPM endpoints ignoring system suspend
by Chancel Liu 09 Oct '23

09 Oct '23

Add a property to list DAPM endpoints which mark paths between these endpoints should not be disabled when system enters in suspend state. LPA means low power audio case. On asymmetric multiprocessor, there are Cortex-A core and Cortex-M core, Linux is running on Cortex-A core, RTOS or other OS is running on Cortex-M core. The audio hardware devices can be controlled by Cortex-M. LPA can be explained as a mechanism that Cortex-A allocates a large buffer and fill audio data, then Cortex-A can enter into suspend for the purpose of power saving. Cortex-M continues to play the sound during suspend phase of Cortex-A. When the data in buffer is consumed, Cortex-M will trigger the Cortex-A to wakeup to fill data. LPA requires some audio paths still enabled when Cortex-A enters into suspend. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 188f38baddec..d8fd17615bf2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -91,6 +91,21 @@ properties: - rpmsg-audio-channel - rpmsg-micfil-channel + fsl,lpa-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of DAPM endpoints which mark paths between these endpoints should + not be disabled when system enters in suspend state. LPA means low power + audio case. On asymmetric multiprocessor, there are Cortex-A core and + Cortex-M core, Linux is running on Cortex-A core, RTOS or other OS is + running on Cortex-M core. The audio hardware devices can be controlled by + Cortex-M. LPA can be explained as a mechanism that Cortex-A allocates a + large buffer and fill audio data, then Cortex-A can enter into suspend + for the purpose of power saving. Cortex-M continues to play the sound + during suspend phase of Cortex-A. When the data in buffer is consumed, + Cortex-M will trigger the Cortex-A to wakeup to fill data. LPA requires + some audio paths still enabled when Cortex-A enters into suspend. + required: - compatible -- 2.25.1

2 6

[RFC PATCH v5 00/11] Add audio support in v4l2 framework
by Shengjiu Wang 09 Oct '23

09 Oct '23

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (11): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Add audio rate controls support media: uapi: define audio sample format fourcc type media: imx-asrc: Add memory to memory driver media: audm2m: add virtual driver for audio memory to memory .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-asrc-rate.rst | 36 + .../userspace-api/media/v4l/pixfmt-audio.rst | 277 ++++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 7 + .../media/videodev2.h.rst.exceptions | 4 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 12 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1251 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 9 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/audm2m.c | 808 +++++++++++ drivers/media/v4l2-core/v4l2-ctrls-core.c | 5 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 4 + drivers/media/v4l2-core/v4l2-dev.c | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 104 ++ include/media/v4l2-ctrls.h | 1 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 58 + include/uapi/linux/v4l2-controls.h | 13 + include/uapi/linux/videodev2.h | 77 + sound/soc/fsl/fsl_asrc.c | 136 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 214 +++ sound/soc/fsl/fsl_easrc.h | 8 +- 35 files changed, 3179 insertions(+), 3 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-asrc-rate.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/audm2m.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

4 18

[PATCH] ALSA: scarlett2: Add Focusrite Clarett+ 2Pre and 4Pre support
by Geoffrey D. Bennett 07 Oct '23

07 Oct '23

The Focusrite Clarett+ series uses the same protocol as the Scarlett Gen 2 and Gen 3 series. This patch adds support for the Clarett+ 2Pre and Clarett+ 4Pre similarly to the existing 8Pre support by adding appropriate entries to the scarlett2 driver. The Clarett 2Pre USB and 4Pre USB presumably use the same protocol as well, so support for them can easily be added if someone can test. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_quirks.c | 2 + sound/usb/mixer_scarlett_gen2.c | 97 ++++++++++++++++++++++++++++++++- 2 files changed, 98 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 9911859d2750..0db94ead1b93 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3421,6 +3421,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */ case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */ case USB_ID(0x1235, 0x8208): /* Focusrite Clarett 8Pre USB */ + case USB_ID(0x1235, 0x820a): /* Focusrite Clarett+ 2Pre */ + case USB_ID(0x1235, 0x820b): /* Focusrite Clarett+ 4Pre */ case USB_ID(0x1235, 0x820c): /* Focusrite Clarett+ 8Pre */ err = snd_scarlett_gen2_init(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 481a80e160de..57e7fdff6905 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -6,7 +6,7 @@ * - 6i6/18i8/18i20 Gen 2 * - Solo/2i2/4i4/8i6/18i8/18i20 Gen 3 * - Clarett 8Pre USB - * - Clarett+ 8Pre + * - Clarett+ 2Pre/4Pre/8Pre * * Copyright (c) 2018-2023 by Geoffrey D. Bennett <g at b4.vu> * Copyright (c) 2020-2021 by Vladimir Sadovnikov <sadko4u(a)gmail.com> @@ -60,6 +60,10 @@ * Support for Clarett USB 8Pre added in Sep 2023 (thanks to Philippe * Perrot for confirmation). * + * Support for Clarett+ 4Pre and 2Pre added in Sep 2023 (thanks to + * Gregory Rozzo for donating a 4Pre, and David Sherwood and Patrice + * Peterson for usbmon output). + * * This ALSA mixer gives access to (model-dependent): * - input, output, mixer-matrix muxes * - mixer-matrix gain stages @@ -832,6 +836,95 @@ static const struct scarlett2_device_info s18i20_gen3_info = { } }, }; +static const struct scarlett2_device_info clarett_2pre_info = { + .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .line_out_hw_vol = 1, + .level_input_count = 2, + .air_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 4 }, + [SCARLETT2_PORT_TYPE_SPDIF] = { 2, 0 }, + [SCARLETT2_PORT_TYPE_ADAT] = { 8, 0 }, + [SCARLETT2_PORT_TYPE_MIX] = { 10, 18 }, + [SCARLETT2_PORT_TYPE_PCM] = { 4, 12 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_PCM, 0, 12 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 8 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 26 }, + { 0, 0, 0 }, + } }, +}; + +static const struct scarlett2_device_info clarett_4pre_info = { + .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .line_out_hw_vol = 1, + .level_input_count = 2, + .air_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 8, 6 }, + [SCARLETT2_PORT_TYPE_SPDIF] = { 2, 2 }, + [SCARLETT2_PORT_TYPE_ADAT] = { 8, 0 }, + [SCARLETT2_PORT_TYPE_MIX] = { 10, 18 }, + [SCARLETT2_PORT_TYPE_PCM] = { 8, 18 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_PCM, 0, 18 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 14 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 12 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 24 }, + { 0, 0, 0 }, + } }, +}; + static const struct scarlett2_device_info clarett_8pre_info = { .config_set = SCARLETT2_CONFIG_SET_CLARETT, .line_out_hw_vol = 1, @@ -907,6 +1000,8 @@ static const struct scarlett2_device_entry scarlett2_devices[] = { /* Supported Clarett USB/Clarett+ devices */ { USB_ID(0x1235, 0x8208), &clarett_8pre_info, "Clarett USB" }, + { USB_ID(0x1235, 0x820a), &clarett_2pre_info, "Clarett+" }, + { USB_ID(0x1235, 0x820b), &clarett_4pre_info, "Clarett+" }, { USB_ID(0x1235, 0x820c), &clarett_8pre_info, "Clarett+" }, /* End of list */ -- 2.41.0

2 2

[PATCH 1/2] ALSA: aloop: Add support for the non-interleaved access mode
by Ivan Orlov 06 Oct '23

06 Oct '23

The current version of the loopback driver supports interleaved access mode only. This patch introduces support for the non-interleaved access mode. When in the interleaved mode, the 'copy_play_buf' function copies data from the playback to the capture buffer using one memcpy call. This call copies samples for multiple, interleaved channels. In the non-interleaved mode we have multiple channel buffers, so we have to perform multiple memcpy calls to copy samples channel after channel. Add new function called 'copy_play_buf_part_n', which copies a part of each channel buffer from playback to capture. Modify the 'copy_play_buf' to use the corresponding memory copy function(just memcpy / copy_play_buf_part_n) depending on the access mode. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/aloop.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a38e602b4fc6..ab116b1fed96 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -158,6 +158,9 @@ struct loopback_pcm { unsigned long last_jiffies; /* If jiffies timer is used */ struct timer_list timer; + + /* size of per channel buffer in case of non-interleaved access */ + unsigned int channel_buf_n; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -335,7 +338,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) substream->runtime; check = runtime->format != cruntime->format || runtime->rate != cruntime->rate || - runtime->channels != cruntime->channels; + runtime->channels != cruntime->channels || + runtime->access != cruntime->access; if (!check) return 0; if (stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -472,6 +476,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) dpcm->buf_pos = 0; dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + dpcm->channel_buf_n = dpcm->pcm_buffer_size / runtime->channels; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { /* clear capture buffer */ dpcm->silent_size = dpcm->pcm_buffer_size; @@ -522,6 +527,22 @@ static void clear_capture_buf(struct loopback_pcm *dpcm, unsigned int bytes) } } +static void copy_play_buf_part_n(struct loopback_pcm *play, struct loopback_pcm *capt, + unsigned int size, unsigned int src_off, unsigned int dst_off) +{ + unsigned int channels = capt->substream->runtime->channels; + unsigned int size_p_ch = size / channels; + unsigned int src_off_ch = src_off / channels; + unsigned int dst_off_ch = dst_off / channels; + int i; + + for (i = 0; i < channels; i++) { + memcpy(capt->substream->runtime->dma_area + capt->channel_buf_n * i + dst_off_ch, + play->substream->runtime->dma_area + play->channel_buf_n * i + src_off_ch, + size_p_ch); + } +} + static void copy_play_buf(struct loopback_pcm *play, struct loopback_pcm *capt, unsigned int bytes) @@ -556,7 +577,11 @@ static void copy_play_buf(struct loopback_pcm *play, size = play->pcm_buffer_size - src_off; if (dst_off + size > capt->pcm_buffer_size) size = capt->pcm_buffer_size - dst_off; - memcpy(dst + dst_off, src + src_off, size); + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) + copy_play_buf_part_n(play, capt, size, src_off, dst_off); + else + memcpy(dst + dst_off, src + src_off, size); capt->silent_size = 0; bytes -= size; if (!bytes) @@ -878,7 +903,7 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NONINTERLEAVED), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | -- 2.34.1

3 6

[PATCH v3 0/4] ALSA: hda: Abstract and update HOST-stream setup procedure
by Cezary Rojewski 06 Oct '23

06 Oct '23

The patchset targets two intertwined topics: The driver shall poll SDxFIFOS to ensure a valid value is set by the controller after programming SDxFMT. Due to amount of users and limited-number of configuration available in our CI when compared to overall possibilities on the market, the check is non-blocking. Second topic relates to stream setup procedure. The procedure differs between HDAudio controller device revisions. Right now those differences are handled directly by a platform driver. Existing top-level 'if (pci->device == APL)' could be replaced by a abstraction in lower parts of the code instead. With that done, the two users are updated accordingly. In avs-driver case, this updates the flow to the recommended one. Changes in v3: - fixed issues pointed out by scripts/kernel-doc Changes in v2: - fixed ->host_setup assignment in patch 02/04 Cezary Rojewski (4): ALSA: hda: Poll SDxFIFOS after programming SDxFMT ALSA: hda: Introduce HOST stream setup mechanism ASoC: Intel: avs: Use helper to setup HOST stream ASoC: Intel: Skylake: Use helper to setup HOST stream include/sound/hda_register.h | 2 ++ include/sound/hdaudio.h | 3 +++ include/sound/hdaudio_ext.h | 3 +++ sound/hda/ext/hdac_ext_stream.c | 41 +++++++++++++++++++++++++++++++ sound/hda/hdac_stream.c | 8 ++++++ sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 14 +---------- 7 files changed, 59 insertions(+), 14 deletions(-) -- 2.25.1

2 8

[PATCH][next] ALSA: 6fire: Fix undefined behavior bug in struct midi_runtime
by Gustavo A. R. Silva 06 Oct '23

06 Oct '23

`struct urb` is a flexible structure, which means that it contains a flexible-array member at the bottom. This could potentially lead to an overwrite of the objects following `out_urb` in `struct midi_runtime`, among them a function pointer. Fix this by placing the declaration of object `out_urb` at the end of `struct midi_runtime`. Fixes: c6d43ba816d1 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/usb/6fire/midi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index 47640c845903..df3f41913274 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -22,12 +22,12 @@ struct midi_runtime { spinlock_t in_lock; spinlock_t out_lock; struct snd_rawmidi_substream *out; - struct urb out_urb; u8 out_serial; /* serial number of out packet */ u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); + struct urb out_urb; }; int usb6fire_midi_init(struct sfire_chip *chip); -- 2.34.1

4 3

[PATCH][next] ALSA: 6fire: Fix undefined behavior bug in struct comm_runtime
by Gustavo A. R. Silva 06 Oct '23

06 Oct '23

`struct urb` is a flexible structure, which means that it contains a flexible-array member at the bottom. This could potentially lead to an overwrite of the objects following `receiver` in `struct comm_runtime`, among them some function pointers. Fix this by placing the declaration of object `receiver` at the end of `struct comm_runtime`. Fixes: ddb6b5a96437 ("ALSA: 6fire: fix DMA issues with URB transfer_buffer usage") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/usb/6fire/comm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 2447d7ecf179..ee81572a4eec 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -19,7 +19,6 @@ enum /* settings for comm */ struct comm_runtime { struct sfire_chip *chip; - struct urb receiver; u8 *receiver_buffer; u8 serial; /* urb serial */ @@ -30,6 +29,7 @@ struct comm_runtime { int (*write8)(struct comm_runtime *rt, u8 request, u8 reg, u8 value); int (*write16)(struct comm_runtime *rt, u8 request, u8 reg, u8 vh, u8 vl); + struct urb receiver; }; int usb6fire_comm_init(struct sfire_chip *chip); -- 2.34.1

4 3

[PATCH] sound: fix name of SIS7013 sound chip in comment
by Thomas Perl 06 Oct '23

06 Oct '23

While grep'ing for SIS7012, I noticed that there is only one reference to it in sound/pci/intel8x0m.c, while most of its code lives in sound/pci/intel8x0.c. This probably was a simple copy'n'paste mistake, as the sound/pci/intel8x0m.c driver implements support for SIS7013 (see DEVICE_SIS), and the two devices seem to share the same behavior / registers. Signed-off-by: Thomas Perl <m(a)thp.io> --- sound/pci/intel8x0m.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 2845cc006d0c..653ecca78238 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -918,7 +918,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) } if (chip->device_type == DEVICE_SIS) { - /* unmute the output on SIS7012 */ + /* unmute the output on SIS7013 */ iputword(chip, 0x4c, igetword(chip, 0x4c) | 1); } -- 2.39.2

2 1

[PATCH v3] ALSA: pcm: oss: Fix race at SNDCTL_DSP_SETTRIGGER
by Ma Ke 06 Oct '23

06 Oct '23

There is a small race window at snd_pcm_oss_set_trigger() that is called from OSS PCM SNDCTL_DSP_SETTRIGGER ioctl; namely the function calls snd_pcm_oss_make_ready() at first, then takes the params_lock mutex for the rest. When the stream is set up again by another thread between them, it leads to inconsistency, and may result in unexpected results such as NULL dereference of OSS buffer as a fuzzer spotted recently. The fix is simply to cover snd_pcm_oss_make_ready() call into the same params_lock mutex with snd_pcm_oss_make_ready_locked() variant. Signed-off-by: Ma Ke <make_ruc2021(a)163.com> --- sound/core/oss/pcm_oss.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 728c211142d1..fd9d23c3684b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2083,21 +2083,16 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; - if (psubstream) { - err = snd_pcm_oss_make_ready(psubstream); - if (err < 0) - return err; - } - if (csubstream) { - err = snd_pcm_oss_make_ready(csubstream); - if (err < 0) - return err; - } if (psubstream) { runtime = psubstream->runtime; cmd = 0; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + err = snd_pcm_oss_make_ready_locked(psubstream); + if (err < 0) { + mutex_unlock(&runtime->oss.params_lock); + return err; + } if (trigger & PCM_ENABLE_OUTPUT) { if (runtime->oss.trigger) goto _skip1; @@ -2128,6 +2123,11 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = 0; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + err = snd_pcm_oss_make_ready_locked(csubstream); + if (err < 0) { + mutex_unlock(&runtime->oss.params_lock); + return err; + } if (trigger & PCM_ENABLE_INPUT) { if (runtime->oss.trigger) goto _skip2; -- 2.37.2

2 1

[PATCH v7 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 05 Oct '23

05 Oct '23

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules dt-bindings: usb: dwc3: Limit num-hc-interrupters definition dt-bindings: usb: xhci: Add num-hc-interrupters definition usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: qcom: qdsp6: Add SND kcontrol to select offload device ASoC: qcom: qdsp6: Add SND kcontrol for fetching offload status ASoC: qcom: qdsp6: Add headphone jack for offload connection status ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../devicetree/bindings/sound/qcom,q6usb.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 4 + .../devicetree/bindings/usb/usb-xhci.yaml | 6 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 102 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 198 ++ drivers/usb/host/xhci.c | 111 +- drivers/usb/host/xhci.h | 105 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 57 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 51 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 56 + sound/soc/qcom/qdsp6/q6afe.c | 206 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 450 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 123 ++ sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 94 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1854 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4896 insertions(+), 196 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb.yaml create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

6 50

[PATCH v2] ASoC: dt-bindings: rt5616: Convert to dtschema
by Bragatheswaran Manickavel 03 Oct '23

03 Oct '23

Convert the rt5616 audio CODEC bindings to DT schema Signed-off-by: Bragatheswaran Manickavel <bragathemanick0908(a)gmail.com> --- V1 -> V2: Changes codec to audio-codec and added spacing above .../bindings/sound/realtek,rt5616.yaml | 49 +++++++++++++++++++ .../devicetree/bindings/sound/rt5616.txt | 32 ------------ 2 files changed, 49 insertions(+), 32 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5616.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml new file mode 100644 index 000000000000..a387f802afd1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5616.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5616 ALSA SoC audio codec driver + +description: | + Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +maintainers: + - Bard Liao <bardliao(a)realtek.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5616 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt deleted file mode 100644 index 540a4bf252e4..000000000000 --- a/Documentation/devicetree/bindings/sound/rt5616.txt +++ /dev/null @@ -1,32 +0,0 @@ -RT5616 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt5616". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC. - -- clock-names: Should be "mclk". - -Pins on the device (for linking into audio routes) for RT5616: - - * IN1P - * IN2P - * IN2N - * LOUTL - * LOUTR - * HPOL - * HPOR - -Example: - -rt5616: codec@1b { - compatible = "realtek,rt5616"; - reg = <0x1b>; -}; -- 2.34.1

3 2

[PATCH] ASoC: SOF: Intel: Add error handling in hda_dsp_trace_release
by liuhaoran 03 Oct '23

03 Oct '23

This patch adds error-handling for the hda_dsp_stream_put() inside the hda_dsp_trace_release function in the hda-trace.c file. Signed-off-by: liuhaoran <liuhaoran14(a)163.com> --- sound/soc/sof/intel/hda-trace.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c index cbb9bd7770e6..5dcd43b9cbe4 100644 --- a/sound/soc/sof/intel/hda-trace.c +++ b/sound/soc/sof/intel/hda-trace.c @@ -73,13 +73,22 @@ int hda_dsp_trace_release(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct hdac_stream *hstream; + int ret; if (hda->dtrace_stream) { hstream = &hda->dtrace_stream->hstream; - hda_dsp_stream_put(sdev, + ret = hda_dsp_stream_put(sdev, SNDRV_PCM_STREAM_CAPTURE, hstream->stream_tag); + hda->dtrace_stream = NULL; + + if (ret < 0) { + dev_dbg(sdev->dev, + "stream put failed: %d\n", ret); + return ret; + } + return 0; } -- 2.17.1

2 1

[PATCH v2 2/2] ASoC: Adds support for TAS575x to the pcm512x driver
by Joerg Schambacher 03 Oct '23

03 Oct '23

Enables the existing pcm512x driver to control the almost compatible TAS5754 and -76 amplifers. Both amplifiers support only an I2C interface and the internal PLL must be always on to provide necessary clocks to the amplifier section. Tested on TAS5756 with support from Andreas Arbesser-Krasser from Texas Instruments <a-krasser(a)ti.com> Signed-off-by: Joerg Schambacher <joerg.hifiberry(a)gmail.com> --- sound/soc/codecs/pcm512x-i2c.c | 4 ++++ sound/soc/codecs/pcm512x.c | 36 +++++++++++++++++++++++++++++++--- 2 files changed, 37 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 5cd2b64b9337..4be476a280e1 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -39,6 +39,8 @@ static const struct i2c_device_id pcm512x_i2c_id[] = { { "pcm5122", }, { "pcm5141", }, { "pcm5142", }, + { "tas5754", }, + { "tas5756", }, { } }; MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); @@ -49,6 +51,8 @@ static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5122", }, { .compatible = "ti,pcm5141", }, { .compatible = "ti,pcm5142", }, + { .compatible = "ti,tas5754", }, + { .compatible = "ti,tas5756", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 89059a673cf0..aa8edf87b743 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -48,6 +48,7 @@ struct pcm512x_priv { int mute; struct mutex mutex; unsigned int bclk_ratio; + int force_pll_on; }; /* @@ -1258,10 +1259,34 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN, - PCM512x_PLLE, 0); + if (!pcm512x->force_pll_on) { + ret = regmap_update_bits(pcm512x->regmap, + PCM512x_PLL_EN, PCM512x_PLLE, 0); + } else { + /* provide minimum PLL config for TAS575x clocking + * and leave PLL enabled + */ + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_0, 0x01); + if (ret != 0) { + dev_err(component->dev, + "Failed to set pll coefficient: %d\n", ret); + return ret; + } + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_COEFF_1, 0x04); + if (ret != 0) { + dev_err(component->dev, + "Failed to set pll coefficient: %d\n", ret); + return ret; + } + ret = regmap_write(pcm512x->regmap, + PCM512x_PLL_EN, 0x01); + dev_dbg(component->dev, "Enabling PLL for TAS575x\n"); + } + if (ret != 0) { - dev_err(component->dev, "Failed to disable pll: %d\n", ret); + dev_err(component->dev, "Failed to set pll mode: %d\n", ret); return ret; } } @@ -1659,6 +1684,11 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) ret = -EINVAL; goto err_pm; } + + if (!strcmp(np->name, "tas5756") || + !strcmp(np->name, "tas5754")) + pcm512x->force_pll_on = 1; + dev_dbg(dev, "Device ID: %s\n", np->name); } #endif -- 2.34.1

2 1

[PATCH v2 1/2] ASoC: pcm512x: Adds bindings for TAS575x devices
by Joerg Schambacher 03 Oct '23

03 Oct '23

The TAS5754/6 power amplifiers use the same pcm512x driver with only minor restictions described in the bindings document. Signed-off-by: Joerg Schambacher <joerg.hifiberry(a)gmail.com> --- Documentation/devicetree/bindings/sound/pcm512x.txt | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index 3aae3b41bd8e..77006a4aec4a 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -1,12 +1,12 @@ -PCM512x audio CODECs +PCM512x and TAS575x audio CODECs/amplifiers These devices support both I2C and SPI (configured with pin strapping -on the board). +on the board). The TAS575x devices only support I2C. Required properties: - - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or - "ti,pcm5142" + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141", + "ti,pcm5142", "ti,tas5754" or "ti,tas5756" - reg : the I2C address of the device for I2C, the chip select number for SPI. @@ -25,6 +25,7 @@ Optional properties: through <6>. The device will be configured for clock input on the given pll-in pin and PLL output on the given pll-out pin. An external connection from the pll-out pin to the SCLK pin is assumed. + Caution: the TAS-desvices only support gpios 1,2 and 3 Examples: -- 2.34.1

3 2