Alsa-devel January 2015

alsa-devel@alsa-project.org

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288 discussions

[alsa-devel] [PATCH v2 1/2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Apr '17

04 Apr '17

From: Zidan Wang <b50113(a)freescale.com> When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 51 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 48 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 75cd7de..fd25973 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,14 +620,38 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - /* Set VMID to 2x50k */ - snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_STANDBY: + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + + /* Set VMID to 2x50k */ + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + break; + + case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + break; + + default: + break; + } + break; case SND_SOC_BIAS_STANDBY: @@ -674,7 +700,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +741,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -1002,6 +1041,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

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[alsa-devel] [PATCH 0/13] ALSA: Deletion of some unnecessary checks
by SF Markus Elfring 26 Oct '16

26 Oct '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 4 Jan 2015 11:50:12 +0100 Further update suggestions were taken into account after several patches were applied from static source code analysis. Markus Elfring (3): seq: Deletion of unnecessary checks before the function call "snd_midi_event_free" oss: Deletion of unnecessary checks before the function call "vfree" emu10k1: Delete an unnecessary check before the function call "kfree" oxygen: Delete an unnecessary check before the function call "snd_pcm_suspend" emux: Delete an unnecessary check before the function call "snd_sf_free" ASoC: Intel: Delete an unnecessary check before the function call "sst_dma_free" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_enable" ASoC: Intel: Delete an unnecessary check before the function call "release_firmware" i2sbus: Delete an unnecessary check before the function call "snd_pcm_suspend_all" sb: Delete an unnecessary check before the function call "snd_emux_free" Deletion of checks before the function call "iounmap" msnd: One function call less in snd_msnd_attach() after error detection msnd: Fix centralized exiting from snd_msnd_attach() sound/aoa/soundbus/i2sbus/core.c | 13 ++++++------- sound/arm/aaci.c | 4 ++-- sound/drivers/ml403-ac97cr.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 31 ++++++++++++++++--------------- sound/parisc/harmony.c | 4 +--- sound/pci/ad1889.c | 5 +---- sound/pci/asihpi/hpioctl.c | 6 ++---- sound/pci/atiixp.c | 3 +-- sound/pci/atiixp_modem.c | 3 +-- sound/pci/aw2/aw2-alsa.c | 4 +--- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 6 ++---- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/ctxfi/cthw20k1.c | 5 +---- sound/pci/ctxfi/cthw20k2.c | 5 +---- sound/pci/echoaudio/echoaudio.c | 6 +----- sound/pci/hda/hda_intel.c | 3 +-- sound/pci/lola/lola.c | 6 ++---- sound/pci/mixart/mixart.c | 7 +++---- sound/pci/nm256/nm256.c | 6 ++---- sound/pci/rme9652/hdsp.c | 4 +--- sound/pci/rme9652/hdspm.c | 4 +--- sound/pci/rme9652/rme9652.c | 3 +-- sound/pci/sis7019.c | 5 +---- sound/pci/ymfpci/ymfpci_main.c | 3 +-- sound/ppc/pmac.c | 15 +++++---------- 26 files changed, 58 insertions(+), 103 deletions(-) -- 2.2.1

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[alsa-devel] Help with wm8731 support for the Utilite (imx6q)
by Jonathan Bennett 06 Apr '16

06 Apr '16

Hey, I'm working with a compulab utilite, trying to get everything working with 3.17, with the eventual goal of getting better support for it upstream. Currently fighting the sound. It's a imx6q talking to a wm8731 over i2c/i2s. I am booting Fedora 21 arm Alpha, with my own dtb ( http://pastebin.com/4a4PBKwx) and driver (http://pastebin.com/1CsesuVV) It shows up in aplay -l, but trying to play anything to it gives me a div by 0 in a driver: [<c04f27e4>] (Ldiv0) from [<bf50512c>] (fsl_ssi_hw_params+0x194/0x364 [snd_soc_fsl_ssi]) This div0 happens in sound/soc/fsl/fsl_ssi.c, in the function fsl_ssi_set_bclk. My driver file is an almost direct copy of what compulab shipped with a 3.10 kernel. This is my first attempt at kernel hacking, so any help would be appreciated. ~Jonathan Bennett

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[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 06 Mar '16

06 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

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[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] 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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Alsa-devel January 2015