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

[alsa-devel] [PATCH V6 0/4] update supported sample format
by Shengjiu Wang 18 Jan '20

18 Jan '20

This patch serial is to update the supported format for fsl_asrc and fix some format issue Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. changes in v5 - free asrc pair when error happens in patch 4/4 changes in v6 - ignore return value of snd_soc_set_runtime_hwparams in patch 4/4 include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 64 ++++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 199 insertions(+), 86 deletions(-) -- 2.21.0

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[alsa-devel] Simultaneous play/record on implicit feedback device causes 'endpoint in use' error
by Robert Giaraffa 15 Jan '20

15 Jan '20

My company (Silicon Labs) makes a CP2615 USB audio class 1.0 device which uses implicit feedback. I saw these two issues when testing with aplay/arecord/alsabat/alsaloop: * During playback-only operation, the implicit feedback (IN) endpoint was not active. * During play/record operation, the implicit feedback values were not acted on, i.e. the host always sent 192 bytes per OUT packet even if previous IN packets had been 188 or 196 bytes. This behavior was observed on these systems: RaspberryPi2: Raspbian 9 (stretch), kernel v4.14.72 x86-64: Ubuntu 18.04, kernel v4.15.0-36 I was able to get implicit feedback working properly in playback mode by adding a device-specific quirk to 'pcm.c' that is similar to existing quirks for implicit-feedback devices, e.g. M-Audio Fast Track Ultra. (Note: The following patch applies to kernel v4.14.72. In v4.18, the set_sync_ep_implicit_fb_quirk function was changed so that there is a single call to usb_ifnum_to_if(), rather than a call for each device-dependent quirk.) $ git diff pcm.c diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b9c9a19f..6fb5567b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -358,6 +358,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x10C4, 0xEAC1): // Silicon Labs CP2615 + // Ensure CP2615 is configured in Async Mode + if (attr != USB_ENDPOINT_SYNC_ASYNC) + return 0; + + // In P16R16 async mode, AudioIN EP (IF:4 EP:0x83) used for implicit feedback + ep = 0x83; + iface = usb_ifnum_to_if(dev, 4); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && After adding the patch, I verified that implicit feedback worked properly for playback, i.e. the OUT samples per packet tracked previous IN values. However, when attempting to perform simultaneous play and record via alsabat, alsaloop, or separate aplay/arecord, the system reports the 'endpoint already in use' error when attempting to set the format for the IN endpoint that is already in use (note that this dmesg log has customized printk messages): [Oct23 10:13] snd_usb_hw_params->set_format() [ +0.000014] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.001654] set_format()->usb_set_interface 3:1 [ +0.000006] Creating new playback data endpoint 03 [ +0.000004] Creating new capture data endpoint 83 <Begin Playback. SYNC endpoint also starts because feedback is implicit> [ +0.000027] snd_usb_pcm_prepare->set_format() [ +0.000005] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.000755] Setting params for EP 03 (type 0, 12 urbs), ret=0 [ +0.000017] Setting params for EP 83 (type 0, 12 urbs), ret=0 [ +0.000004] Starting DATA_EP 03 @b9224000 [ +0.000004] snd_usb_endpoint_implicit_feedback_sink (03) is TRUE [ +0.001495] Starting SYNC_EP 83 @b9222000 <After playing for 0.5s, start recording. Failure occurs because EP83 is already in use.> [ +0.493035] snd_usb_hw_params->set_format() [ +0.000016] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 [ +0.001549] set_format()->usb_set_interface 4:1 [ +0.000005] Re-using EP 83 in iface 4:1 [ +0.000028] snd_usb_pcm_prepare->set_format() [ +0.000004] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 *** [ +0.003196] *** snd_usb_endpoint_set_params(): EP #83: already in use [ +0.000003] configure_endpoint->endpoint_set_params: err=FFFFFFF0 I bought a used Fast Track Ultra device for comparison testing, and found that it behaved the same way as the CP2615 does with the implicit feedback quirk: in playback-only mode implicit feedback works properly, but play/record causes the 'EP already in use' error. Are there any suggestions on how I can resolve this issue? Any help would be greatly appreciated! Thanks, Bob Giaraffa

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[alsa-devel] [PATCH v6 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 10 Jan '20

10 Jan '20

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v6: - Only define the function when CONFIG_ACPI is set. v5: - Add wording suggestion from Bjorn. v4: - Let caller to find its upstream port device. drivers/pci/pci.c | 18 ++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 20 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..1df99d9e350e 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,24 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +#ifdef CONFIG_ACPI +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); +#endif + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

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[alsa-devel] [PATCH v4 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 02 Jan '20

02 Jan '20

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v4 changes: 1. Fixed wrong signed-off as correct mail suffix. 2. Fixed patch subject that add patch version. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. Change the name of mtk_wdt_compatible to mtk_wdt_data. 3. Remove toprgu_reset struct and use mtk_wdt_dev instead. 4. Get the value of sw_rst_num from .h file. 5. Adddd mt2712-resets.h for mt2712. 6. Improve commit message. v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH v2 0/2] ALSA: usb-audio: fix PCM device order
by Takashi Iwai 30 Dec '19

30 Dec '19

Hi, this is a rewrite of the old USB-audio fix patch I obviously overlooked to adapt to 5.3 kernel with a preliminary cleanup patch. Totally untested, so any test / feedback would be appreciated. thanks, Takashi === Alexander Tsoy (1): ALSA: usb-audio: fix PCM device order Takashi Iwai (1): ALSA: usb-audio: Unify audioformat release code sound/usb/stream.c | 64 +++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 47 insertions(+), 17 deletions(-) -- 2.16.4

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Re: [alsa-devel] snd-bebob : from kernel 4.13 to 5.3.19 and .20
by Takashi Sakamoto 05 Dec '19

05 Dec '19

Hi, On Wed, Oct 30, 2019 at 10:40:18AM +0100, Jean-Paul Argudo wrote: > > I'll install Eoan kernel and test my devices. > > (but I don't have Saffire and Saffire LE...) I'd like you to wait > > for my test report. > > What I can say is that with kernel 4.13 under Ubunto Disco everything > was fine :-( I install Eoan (linux-image-5.3.0-19-generic:5.3.0-19.20) and test ALSA bebob driver with my test devices: - Yamaha GO46 (DM1000 ASIC) - M-Audio FireWire Solo (DM1000 ASIC) - Behringer FCA610 (DM1500 ASIC) - Focusrite SaffirePro 26 I/O (DM1000E) However I don't find similar discontinuities. This means that you encounter model-specific issue as regression I didn't realized. For my information, would you please get output from nodes on procfs? * /proc/asound/cardX/firewire/clock * /proc/asound/cardX/firewire/firmware * /proc/asound/cardX/firewire/formation Additionally, please install 'gir1.2-hinawa-2.0' package, then clone below remote repository: * https://github.com/takaswie/hinawa-utils This repository includes two scripts to get information from the device: * hinawa-bebob-parser * hinawa-bebob-connection-cui When your device is detected as sound card '2' and your device is also detected as '/dev/fw1', execute below: $ ./hinawa-bebob-connection-cui 2 graph-connections $ ./hinawa-bebob-parser 1 2 > At startup it lights green ok, but no sound is playable, then the > lights turn orange (like it is when it's not working), I hear a > "relay sound" (a electric clic of a relay), then, the Saffire LE > disapears from the sound menu in Ubuntu sound menu. Once your device is detected and ALSA bebob driver bound to it, pulseaudio detects it and start PCM substream to get capabilities of the device. In your case, the PCM substream is corrupted by the discontinuity. Then pulseaudio give up to use the device. As a result, sound menu drops entry of the device. I guess that the color of LED corresponds to the packet streaming. On Wed, Oct 30, 2019 at 11:12:01AM +0100, Jean-Paul Argudo wrote: > But something new and maybe interesting for you : > > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa- sink.c: Most likely this is a bug in the ALSA driver 'snd_bebob'. Please report this issue to the ALSA developers. > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Practically these logs can be ignored. (I work to suppress the above lines in development period for Linux kernel v5.5.) > **And in kern.log** : at the beggining in the boot sequence (?), > between FS-cache lines and usb ones: > > Oct 27 21:54:36 deiphobe kernel: [ 8.742105] FS-Cache: N-key=[8] '020001bdc0a800fe' > > Oct 27 21:54:40 deiphobe kernel: [ 12.768168] usb 3-4.1: reset high- speed USB device number 5 using xhci_hcd > Oct 27 21:54:45 deiphobe kernel: [ 18.294126] snd-bebob fw1.0: transaction failed: no ack > Oct 27 21:54:45 deiphobe kernel: [ 18.294167] snd-bebob fw1.0: fail to get section type for isoc out plug 0: -5 > Oct 27 21:54:45 deiphobe kernel: [ 18.294172] snd-bebob fw1.0: fail to run AMDTP slave stream:-5 > Oct 27 21:54:45 deiphobe kernel: [ 18.306399] firewire_ohci 0000:05:01.0: DMA context still active (0x00000411) > Oct 27 21:54:45 deiphobe kernel: [ 18.318425] firewire_ohci 0000:05:01.0: DMA context still active (0x00000411) > Oct 27 21:54:47 deiphobe kernel: [ 20.311440] irq_handler: 20 callbacks suppressed > Oct 27 21:54:47 deiphobe kernel: [ 20.311441] firewire_ohci 0000:05:01.0: isochronous cycle inconsistent > Oct 27 21:54:48 deiphobe kernel: [ 21.322730] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 > > Oct 27 21:54:50 deiphobe kernel: [ 22.970251] rfkill: input handler disabled > Oct 27 20:55:10 deiphobe kernel: [ 42.697284] usb 3-4.1: reset high- speed USB device number 5 using xhci_hcd IEEE 1394 bus 'bus-reset' state. Just before/after the bus-reset, any functionality easily fails. It's better to plug-in the device enough after bootup. Regards Takashi Sakamoto

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[alsa-devel] Surround speaker connection on Acer 8951G
by Sergey 'Jin' Bostandzhyan 28 Nov '19

28 Nov '19

Hi, I am not sure if this is the right place to ask or if I should just submit a bugreport, but you guys helped me out with a similar problem about 10 years ago, so I'll try here first. If I'm wrong here, then I apologize, just let me know. I have an Acer Aspire Ethos 8951G notebook which has 6 built in speakers for 5.1 surround sound, it's a Realtek ALC669X. The problem is that only two speakers are recognized by default. I tried fiddling around with hdajackretast and it does make a difference, but I somehow could not find a step-by-step approach on how to figure out what is what and how to map everything correctly, so some guidance would be very much appreciated. I even ended up in situations where I got channels supporting only mute which would influence muting of other channels, I really got lost there.. Here's my alsa-info output: http://alsa-project.org/db/?f=3adde87164f5fc349c3c78b211ee63e416ebf10b It would be great if we could figure it out and make it work by default. Basically that's what we did 10 years ago, back then I had an Acer Aspire 8920G with a ALC889 which had the same problem that got fixed by you and these days it just works out of the box :) Kind regards, Jin

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[alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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[alsa-devel] parameter for pulse device?
by frederik＠ofb.net 16 Nov '19

16 Nov '19

Dear ALSA, In my ~/.asoundrc, I have lines like this: pcm.mic { type pulse; device "mic" } pcm.monitor { type pulse; device "monitor"; } pcm.music { type pulse; device "music"; } to make it easier for ALSA-aware programs to input and output via PulseAudio, e.g.: ecasound -i alsa,mic -o alsa,monitor -etd:... However, I would like to simplify this and not have to update ~/.asoundrc every time I create a new PulseAudio device. Since ALSA has the ability for PCMs to take a parameter, I thought this might work with the "pulse" PCM and the PulseAudio device name. But I get an "error: Invalid argument" when trying to pass the device name as an argument to the "pulse" PCM: $ ecasound -o alsa,pulse:music -i some.wav ... ALSA lib conf.c:5014:(snd_config_expand) Unknown parameters music ALSA lib pcm.c:2564:(snd_pcm_open_noupdate) Unknown PCM pulse:music ERROR: Connecting chainsetup failed: "Enabling chainsetup: AUDIOIO-ALSA: ... Unable to open ALSA-device for playback; error: Invalid argument" Is there some magic with macros that I can use to accomplish this syntax, or can we add the ability for the "pulse" PCM to take a parameter naming the device? Thanks, Frederick

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[alsa-devel] [PATCH 0/4] soundwire: update ASoC interfaces
by Pierre-Louis Bossart 14 Nov '19

14 Nov '19

We need new fields in existing structures to a) deal with race conditions on codec probe/enumeration b) allow for multi-step ACPI scan/probe/startup on Intel plaforms To avoid conflicts between ASoC and Soundwire trees, these 4 patches are provided out-of-order, before the functionality enabled in these header files is added in follow-up patch series which can be applied separately in the ASoC and Soundwire trees (of course after Vinod and Mark sync-up so that these patches are present in both trees). Pierre-Louis Bossart (3): soundwire: sdw_slave: add new fields to track probe status soundwire: add enumeration_complete structure soundwire: intel: update interfaces between ASoC and SoundWire Rander Wang (1): soundwire: intel: update stream callbacks for hwparams/free stream operations drivers/soundwire/intel.c | 20 ++++-- drivers/soundwire/intel.h | 13 ++-- drivers/soundwire/intel_init.c | 31 +++------ include/linux/soundwire/sdw.h | 7 ++ include/linux/soundwire/sdw_intel.h | 103 +++++++++++++++++++++++++--- 5 files changed, 131 insertions(+), 43 deletions(-) -- 2.20.1

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