January 2022 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 29 Oct '25

29 Oct '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

6 7

[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

2 2

[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

2 23

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

3 5

[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…

3 10

[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

5 18

[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

4 4

[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

3 9

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

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