March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

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[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 13 Nov '15

13 Nov '15

Version 0.2 Users doc: All You need to get a HDMI- Pulseaudio Server on Your sunxi board to output Multichannel PCM is this line in default.pa, ttable pcms in asound.conf get ignored: +load-module module-alsa-sink device="hw:1,0" channels=8 channel_map="front-left,front-right,lfe,front-center,rear-left,rear-right,side-left,side-right" ### Automatically load driver modules depending on the hardware available .ifexists module-udev-detect.so load-module module-udev-detect Probe with speaker-test, You may need other channel mapping than I. If you run PA in system mode please read the security documentation before, see PA log for link. y tom

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[alsa-devel] No stream timing info with alsa pulse plugin?
by sqweek 24 Sep '15

24 Sep '15

Hi guys, I discovered on my machine that alsa_snd_pcm_status_get_tstamp (and also get_htstamp) stop working when my asoundrc invokes the pulse plugin. That is, normally they work fine but once the sound is going via pulse the returned time stamp is always zero. I'm trying to understand why that is. I couldn't really find any discussion of the pulse plugin on the archives so I'm not entirely sure I've found the right list; I was directed here via the pulseaudio bugzilla[1]. [1] https://bugs.freedesktop.org/show_bug.cgi?id=88503 Thanks for any assistance! -sqweek

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[alsa-devel] Intel SST on a Bay Trail tablet
by Antonio Ospite 18 Sep '15

18 Sep '15

Hi, I am trying to get the Intel SST driver working on a Teclast X98 Air 3G [1], it's a Bay Trail tablet. The tests below have been made with 4.0.0-rc1 and the firmware files from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The DSDT on this device is quite broken, and I am patching it to get stuff working. The audio device works just fine under Android, I don't know if that's because the Android kernel is ignoring the DSDT and is hard-coding some platform data. Anyhow, with the untouched DSDT[2] I was getting this message in dmesg[3]: [ 11.344095] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 11.404831] resource sanity check: requesting [mem 0x91000000-0x911fffff], which spans more than ADMA0F28:00 [mem 0x91000000-0x91000fff] [ 11.404833] ------------[ cut here ]------------ [ 11.404849] WARNING: CPU: 0 PID: 213 at arch/x86/mm/ioremap.c:197 __ioremap_caller+0x22c/0x350() ... I guessed this was because the DSDT defined an ADMA0F28 device with a memory resource overlapping with the LPMA device. So I removed the overlapping resources (ADMA0F28 and SSPX0000) in the DSDT and then the driver attempted to boot the audio device but failed with this message in dmesg[4]: [ 11.602291] baytrail-pcm-audio baytrail-pcm-audio: ipc: error DSP boot timeout ... Then I noticed that the Interrupt resources of the LPMA device were out of order in the DSDT, so I put the Interrupt 0x18 as the first one[5] and now the driver can boot the device and upload the firmware to it[6]: [ 11.433460] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 11.433468] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 11.433473] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 11.448257] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok ... But still no sound. I can see the mixer controls in alsamixer and I tried to initialize them copying from the alsactl output from the Android kernel[7], but no, still no sound. I do not know if that matters, but the driver still emits this message: [ 11.398304] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 11.398314] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 ... I noticed that on the Android kernel[8] the DMA channels for the ADMA0F28 device are actually configured, the intel_mid_dma driver sets up 4 channels for it along with 2 x 6 channels for the INTL9C60 device. On the mainline kernel only the INTL9C60 device gets configured, by the dw_dmac driver and with 2 x 8 channels. I do not know much about these details tho. Also, the codec driver on the Android kernel (byt_rt5642) seems to use one GPIO as an interrupt[9]: 390: 0 1 0 0 VLV-GPIO-gpio byt-codec-int Which is pin 4 on GPO2, but the codec driver in the mainline kernel does not seem to be using this; however AFAICS in sound/soc/codecs/rt5640.c a gpio pin in only used for suspend/resume, so it is not strictly needed if it's already in the correct state, right? Last question, what is the difference between having the device detected by sound/soc/intel/sst-acpi.c (like in my case) opposed to sound/soc/intel/sst/sst_acpi.c? I can see the 80860F28 id in both the files but they seem to load different firmwares. Any idea of what is missing? It's the DSDT which still needs adjusting, or are some changes needed (also) in the kernel driver? Thanks, Antonio [1] http://ao2.it/110 [2] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_original.dsl [3] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_ori… [4] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [5] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_without_ADMA0… [6] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [7] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_asound.state [8] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_Android_Teclast_… [9] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_proc_interrupt… -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

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[alsa-devel] Possible GPL-licensed header included in alsa-lib
by Clemens Lang 27 Jul '15

27 Jul '15

Hi, I recently reviewed the alsa-lib 1.0.27.2 source code from a licensing perspective to ensure compliant usage and distribution. It seems the intention is for the libraries to be covered by LGPL. I had a look at include/list.h, which seems to be licensed under GPL-2.0. It does not have a license header or any other copyright information, but it states it was taken from Linux 2.4.0: include/list.h, line 5: | This code was taken from the Linux 2.4.0 kernel. [jaroslav] I'm not sure one could argue that the header was trivial, especially since it does not only contain constants, structures and interface definitions, but also implementations of inline functions. I found one previous inquiry about a similar issue[1], but it does not cover the include/list.h header. Two questions: 1. Have the copyright holders of the code in question agreed to re-license under a different, LGPL-preserving license and, if yes, is there some documentation about that? 2. If the header is indeed covered by GPL-2.0, would you accept patches to replace it with a different implementation, e.g. Rusty Russell's MIT-licensed list from ccan[2]? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/038267.html [2] http://ccodearchive.net/info/list.html Thank you for your time, Clemens Lang -- BMW Car IT GmbH Clemens Lang Spezialist Entwicklung Lise-Meitner-Straße 14 89081 Ulm Mail: clemens.lang(a)bmw-carit.de Web: http://www.bmw-carit.de -------------------------------------------------------------------- BMW Car IT GmbH Geschäftsführer: Michael Würtenberger und Reinhard Stolle Sitz und Registergericht: München HRB 134810 --------------------------------------------------------------------

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Re: [alsa-devel] ASoC: Add max98925 codec driver
by Dan Carpenter 20 Jul '15

20 Jul '15

Hello Anish Kumar, The patch 1ff2765182d1: "ASoC: Add max98925 codec driver" from Mar 9, 2015, leads to the following static checker warning: sound/soc/codecs/max98925.c:274 max98925_rate_value() error: buffer overflow 'rate_table' 9 <= 9 sound/soc/codecs/max98925.c 259 static inline int max98925_rate_value(struct snd_soc_codec *codec, 260 int rate, int clock, int *value, int *n, int *m) 261 { 262 int ret = -EINVAL; 263 int i; 264 265 for (i = 0; i < ARRAY_SIZE(rate_table); i++) { 266 if (rate_table[i].rate >= rate) { 267 *value = rate_table[i].sr; 268 *n = rate_table[i].divisors[clock][0]; 269 *m = rate_table[i].divisors[clock][1]; 270 ret = 0; 271 break; 272 } 273 } 274 dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", 275 __func__, rate_table[i].rate, *value); ^^^^^^^^^^^^^^^^^^ If we don't break then we're off by one. 276 return ret; 277 } regards, dan carpenter

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[alsa-devel] [V2 PATCH] ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
by Hui Wang 01 Jul '15

01 Jul '15

Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, they all belong to the cxt5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. This design refers to the similar solution for the realtek codec ALC269 family in the patch_realtek.c. Cc: Alex Hung <alex.hung(a)canonical.com> Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1..1f2717f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3244,9 +3244,29 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} -- 1.8.1.2

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[alsa-devel] Internal speakers not working on HP ProBook 455 G2
by Michel Dänzer 23 Jun '15

23 Jun '15

Hi, the built-in speakers of my HP ProBook 455 G2 are not working by default. I'm attaching the output of alsa-info.sh. In pavucontrol, the speakers appear as "Speakers (unavailable)", and in the GNOME sound settings they don't appear at all. After some playing around with HDAAnalyzer, I discovered that enabling the 'OUT' widget control of pin 0x14 makes the speakers emit sound, see the diff below. Let me know if you need any more information, and I'll happily test any patches to fix this problem. Diff for codec 1/0 (0x10ec0282): --- +++ @@ -120,39 +120,40 @@ bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 - Control: name="Internal Mic Phantom Jack", index=0, device=0 + Control: iface="card", name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60140: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 - Control: name="Speaker Phantom Jack", index=0, device=0 + Control: iface="card", name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE + Pin-ctls: 0x40: OUT Unsolicited: tag=0x00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute= -- Earthling Michel Dänzer | http://www.amd.com Libre software enthusiast | Mesa and X developer

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