June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Codec VIA VT2020/VT1718S and questions about commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45
by alex dot baldacchino dot alsasub at gmail dot com 17 Jun '11

17 Jun '11

Hello, I've been looking into the code of alsa drivers (mainly hda and patch_via.c), with help by Raymond Yau (many thanks to him) after a problem making my system work with a sound driver newer than version 1.0.21 (at last, I fixed it by recompiling alsa-lib to use device files in /dev as created by udev with correct minor numbers - original subject of that discussion on alsa-devel mailing list is "Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)"). My system is a no-brand desktop, with an asrock motherbord (model 890GX Extreme3) embedding an audio codec VIA VT2020 (beside ATI HDMI). That should support 7.1 speaker systems, plus eventually a headset connected to a front audio panel in multistream mode (since VT1718S seems to support as much as 10 playback channles, as per channels_max field in static const struct hda_pcm_stream vt1718S_pcm_analog_playback, file patch_via.c). Though, my motherboard holds no grey (side) jack (would be nid 0x27), and it would seem that blue (line-in) jack (0x2a) should be retasked as side output (it would use AOW 0x0c, actually used for Independent HP mode, but I've tested, with help by Raymond Yau, and can confirm 0x0b can be used instead by front panel line-out - it's in the connection list for nid 0x34, labeled "Independent HP", which connects to 0x1b and thus to 0x28, "Headphone Playback Switch"). We were trying to fix it (most of the code, about all the code about that, coming from him, I'm just testing/reviewing it and looking into some other, perhaps more general and maybe minor, issues). But now we'd have a concern about a piece of code within struct hda_verb vt1718S_volume_init_verbs[], as modified by you in commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45, specifically the following part: @@ -4264,7 +4265,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, My system (codec vt2020) has only 5 connections to nid 0x21: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 and that seems to be the same for other vt1718s codec implementations (by asrock): https://launchpadlibrarian.net/70606587/Card0.Codecs.codec.0.txt (motherboard ASRock H55M Pro, more info at https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/772638 ), therefore your code seems to access a non-existing connection. Since such looks like not very safe/correct, Raymond Yau had suggested to change it into: - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, which seems to work (or at least to cause no problem) on my system. However, after finding your commit, we wonder if that could be necessary for the machine (e.g. because of internal functioning, I'm _not_ asking you to disclose, of course), so I'm writing you to ask for some clarification, as far as possible. Moreover, I have some doubts about the rest of that patch: @@ -4255,7 +4255,8 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Enable MW0 adjust Gain 5 */ + {0x1, 0xfb2, 0x10}, /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ In my system, MW0 (first audio mixer widget) seems to be nid 0x18 (0x1 being AFG Function Id) and I haven't been able to find infos about verb 0xfb2; commenting it out seems to cause no problem, however that may look like an implementation-specific verb to be sent to a special connection (as a part of VIA internals I'll never ask you to disclose), so I'll be grateful to you if you can just tell me if that's specifically needed and must be there. Furthermore: /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, (this part not created by that commit of yours) My system seems to have only 6 Audio Mixer Widgets (0x18, 0x19, 0x1a, 0x1b, 0x1c and 0x21 - unless any Vendor Defined Widget is taken into account), as it seems to be for vt1718S codec as described in links above; in both cases 0x28 has one only connection: vt2020 (mine): Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=21, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1b vt1718s (linked above): Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=05, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1b Thus, it would seem (to me, at least) more natural (and correct) to initialize it by selecting connection 0 (the only existing one - or none at all, since the only one should be selected by the hardware by default), unless there's some specific reason for that (e.g. other codec implementations with more than one connection to 0x28, or internal need for that). Could there be the need to create different initialization verb arrays for different flavours of VT1718S? Many thanks in advance for your answers. Best regards.

2 1

Re: [alsa-devel] Driver for TerraTec DMX 6Fire USB
by Takashi Iwai 16 Jun '11

16 Jun '11

At Tue, 18 Jan 2011 14:41:47 +0100, torsten.schenk wrote: > > Hello, > > (to all those who already got this mail: sorry, I sent the attachment accidentally in html format so it was ignored by alsa-devel ML distributor) > > i recently got a TerraTec DMX 6Fire USB. Since it's not supported by alsa, I decided to look into communications between the windows driver and the card. Here is the result. > > What should be revised: > - I was unsure about naming conventions, so please check the name of card/midi/pcm/module. > - Also the device needs firmware to be able to run, so I included the firmware from the windows driver in the attachment. I read about that Linus Torvalds doesn't like .ihex-parsing in drivers, but I cannot convert the files easily into a binary format because the offsets are not contiguous. Of course there could be alternate ways to be thought about, which would imply that original files from TT are not supported since they are in ihex. > - I didn't manage to create a configuration in /usr/share/alsa/cards. > > What is working: Everything except SPDIF > - Hardware Master volume > - PCM 44-192kHz@24 bits, 6 channels out, 4 channels in (analog) > - MIDI in/out > - firmware loading for coldplugged devices > - phono/line switching > > If anything else comes to your attention, just let me know. Sorry for the late reply. It took some time until I have free slot. First off, there are lots of coding-style issues to be fixed. Try to run scripts/checkpatch.pl once with your patch. Not all warnings have to be fixed (e.g. 80-chars) but would be nice if fixed. Here are some review comments through a very quirk glance: > +static int __devinit probe(struct usb_interface *intf, const struct usb_device_id *usb_id) Adding a short prefix (e.g. usb6fire_probe()) is better in general. Otherwise the stack trace would be difficult to read. Of course, too long names are also annoying, so it's a matter of taste. ... > + chip = (struct sfire_chip*)card->private_data; You can avoid cast here. > + chips[regidx] = chip; > + chip->dev = device; > + chip->regidx = regidx; > + chip->intf_count = 1; > + chip->card = card; > + chip->shutdown = FALSE; > + chip->midi = NULL; > + chip->comm = NULL; > + chip->control = NULL; > + chip->pcm = NULL; NULLs are automatically filled, so these can be skipped. > + ret = snd_usb_6fire_comm_init(chip); > + if(ret < 0) > + { > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_midi_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_pcm_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_control_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_card_register(card); > + if(ret < 0) > + { > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_usb_6fire_control_destroy(chip); > + snd_card_free(card); A general way for error-handling is either using goto or using a common destructor. In either way, you can call the *_free or *_destroy with the non-NULL check of each instance. It's a cleaner in general. > +static void disconnect(struct usb_interface *intf) > +{ > + struct sfire_chip *chip; > + struct snd_card *card; > + > + chip = usb_get_intfdata(intf); > + if(chip) //if !chip, fw upload has been performed so that we don't have a chip. > + { > + card = chip->card; > + chip->intf_count--; > + if (chip->intf_count == 0) > + { > + chip->shutdown = TRUE; > + snd_card_disconnect(card); > + snd_usb_6fire_comm_destroy(chip); > + snd_usb_6fire_control_destroy(chip); > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_card_free(card); > + usb_set_intfdata(intf, NULL); The same destructor as in probe can be used here. > +static struct usb_device_id device_table[] = > +{ > + { > + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, > + .idVendor = 0x0ccd, > + .idProduct = 0x0080 > + }, > + { } > +}; Usually MODULE_DEVICE_TABLE() follows just after the definition. > diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h > --- a/sound/usb/6fire/chip.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/chip.h 2011-01-18 13:58:06.000000000 +0100 > @@ -0,0 +1,30 @@ > +/* > + * Linux driver for TerraTec DMX 6Fire USB > + * > + * Author: Torsten Schenk > + * Created: Jan 01, 2011 > + * Copyright: (C) Torsten Schenk > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + */ > +#pragma once Don't use prgram. Also, add #ifdef xxx for protecting the double inclusion. > +#include "common.h" > + > +struct sfire_chip > +{ > + struct usb_device *dev; > + struct snd_card *card; > + int intf_count; //number of registered interfaces > + int regidx; //index in module parameter arrays > + BOOL shutdown; Don't use own bool type. Use the generic "bool" type, and use true/false instead of TRUE/FALSE (unless the size really matters). > diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c > --- a/sound/usb/6fire/comm.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/comm.c 2011-01-18 13:58:21.000000000 +0100 ... > +static int write_02(struct comm_runtime *rt, __u8 reg, __u16 value) Use u8 and u16. > diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h > --- a/sound/usb/6fire/comm.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/comm.h 2011-01-18 13:58:29.000000000 +0100 > @@ -0,0 +1,42 @@ > +/* > + * Linux driver for TerraTec DMX 6Fire USB > + * > + * Author: Torsten Schenk > + * Created: Jan 01, 2011 > + * Copyright: (C) Torsten Schenk > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; either version 2 of the License, or > + * (at your option) any later version. > + */ > + > +#pragma once Ditto. No pragma and add ifdef. > diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h > --- a/sound/usb/6fire/common.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/common.h 2011-01-18 13:58:38.000000000 +0100 > +#pragma once > + > +#include <linux/slab.h> > +#include <linux/usb.h> > +#include <sound/core.h> > + > +#define BOOL char > +#define TRUE 1 > +#define FALSE 0 Don't use these. > +#define MIN(X, Y) ((X) < (Y) ? (X) : (Y)) > +#define MAX(X, Y) ((X) > (Y) ? (X) : (Y)) Use standard min() / max() macros. > +#define ERR KERN_ERR "6fire: " > +#define WARNING KERN_WARNING "6fire: " Hmm... These are also better to be avoided. At most, define "6fire: " as PREFIX or PFX or whatever. But KERN_ERR should be put plainly. > diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c > --- a/sound/usb/6fire/control.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/control.c 2011-01-18 13:58:57.000000000 +0100 > diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c > --- a/sound/usb/6fire/firmware.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/firmware.c 2011-01-18 13:57:15.000000000 +0100 ... > +static inline int fpga_write(struct usb_device *device, char *data, int len) Avoid inline unless you really need it. Also add MODULE_FIRMWARE() definitions in this file. > diff -Nur a/sound/usb/6fire/Makefile b/sound/usb/6fire/Makefile > --- a/sound/usb/6fire/Makefile 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/Makefile 2011-01-16 23:14:22.000000000 +0100 > @@ -0,0 +1,13 @@ > +obj-m += snd-usb-6fire.o > +snd-usb-6fire-objs += chip.o comm.o midi.o control.o firmware.o pcm.o > + > +all: > + make CONFIG_DEBUG_SECTION_MISMATCH=y -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules > + > +clean: > + make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean > + > +install: > + mkdir -p /lib/modules/$(shell uname -r)/extras > + cp snd-usb-6fire.ko /lib/modules/$(shell uname -r)/extras Remove all, clean, install things. > diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c > --- a/sound/usb/6fire/pcm.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/pcm.c 2011-01-18 13:59:57.000000000 +0100 > diff -Nur a/sound/usb/Kconfig b/sound/usb/Kconfig > --- a/sound/usb/Kconfig 2011-01-16 23:20:41.000000000 +0100 > +++ b/sound/usb/Kconfig 2011-01-16 23:14:22.000000000 +0100 > @@ -32,6 +32,17 @@ > To compile this driver as a module, choose M here: the module > will be called snd-ua101. > +config SND_USB_6FIRE > + tristate "TerraTec DMX 6Fire USB" > + depends on X86 > + select SND_HWDEP Do you really need hwdep? Rather select FW_LOADER, if any. > + select SND_PCM Missing SND_RAWMIDI dependency. > @@ -65,7 +76,6 @@ > * Native Instruments Guitar Rig Session I/O > * Native Instruments Guitar Rig mobile > * Native Instruments Traktor Kontrol X1 > - * Native Instruments Traktor Kontrol S4 > To compile this driver as a module, choose M here: the module > will be called snd-usb-caiaq. > @@ -83,7 +93,6 @@ > * Native Instruments Kore Controller > * Native Instruments Kore Controller 2 > * Native Instruments Audio Kontrol 1 > - * Native Instruments Traktor Kontrol S4 Don't bother others ;) thanks, Takashi

2 20

[alsa-devel] [PATCH] sound: fix lola section mismatch warnings
by Randy Dunlap 16 Jun '11

16 Jun '11

From: Randy Dunlap <randy.dunlap(a)oracle.com> Fix section mismatch warnings in lola sound driver: WARNING: sound/pci/lola/snd-lola.o(.text+0xda4): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_pcm() The function lola_parse_tree() references the function __devinit lola_init_pcm(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_pcm is wrong. WARNING: sound/pci/lola/snd-lola.o(.text+0xdf5): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_pins() The function lola_parse_tree() references the function __devinit lola_init_pins(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_pins is wrong. WARNING: sound/pci/lola/snd-lola.o(.text+0xe52): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_clock_widget() The function lola_parse_tree() references the function __devinit lola_init_clock_widget(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_clock_widget is wrong. Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-30-rc3.orig/sound/pci/lola/lola.c +++ lnx-30-rc3/sound/pci/lola/lola.c @@ -445,7 +445,7 @@ static void lola_reset_setups(struct lol lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ } -static int lola_parse_tree(struct lola *chip) +static int __devinit lola_parse_tree(struct lola *chip) { unsigned int val; int nid, err;

2 1

Re: [alsa-devel] Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)
by alex dot baldacchino dot alsasub at gmail dot com 16 Jun '11

16 Jun '11

(Sorry for previous message being too long) 2011/6/14 Raymond Yau <superquad.vortex2(a)gmail.com>: - Nascondi testo citato - > 2011/6/8 alex dot baldacchino dot alsasub at gmail dot com > <alex.baldacchino.alsasub(a)gmail.com>: >> >> >> http://pastebin.ca/2076750 >> >> >> By the way, should something like the above be done for capture >> streams as well? I've noticed (with front panel disabled) that >> controls "Front Mic Playback Volume" and "Front Mic Playback Switch" >> are correctly missing from audio-mixer widget 0x21, but node 0x29 is >> still 'attached' to control "Front Mic Boost Capture Volume", which is >> accessible through a mixer (though I didn't try to play with it; it is >> listed also by amixer output in above linked document). Is this the >> wanted/expected behaviour? > > It seem that 0x2b and 0x29 does not have any mic boost capture volume, > that look like "Playback Switch" when they are retasked as output for > 3 jacks motherboard > > Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Front Mic Boost Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000233c: IN OUT HP Detect > Vref caps: HIZ 50 100 > Pin Default 0x02a19037: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x7 > Pin-ctls: 0x21: IN VREF_50 > Unsolicited: tag=20, enabled=1 > Power states: D0 D1 D2 D3 > Power: setting=D3, actual=D3 > Connection: 1 > 0x1c > > Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Mic Boost Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00002334: IN OUT Detect > Vref caps: HIZ 50 100 > Pin Default 0x01a19036: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x6 > Pin-ctls: 0x21: IN VREF_50 > Unsolicited: tag=20, enabled=1 > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x0a* 0x0c > That pastebin was alsa-info.sh output with front panel disabled in bios, to show via_build_pcms() can be fixed to avoid creating a sencond playback pcm substream (thus, a second subdevice) when there's no front audio panel in the system (or is disabled), even if snd_hda_find_mixer_ctl() cannot find control "Independent HP" if called from within via_build_pcms: APLAY **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 But I might have pasted a wrong file (sorry for that), correct output for that (sort of) patch would be: Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Front Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x42a190f7: [N/A] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 1 0x1c Front audio panel not connected (not seen by the BIOS) -> front mic not available -> power setting D3 (because of set_pin_power_state() checking for presence, I guess) Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 2 0x0a* 0x0c Rear jack present (on the motherboard), but nothing connected to it -> power state D3 > /* capture mixer elements */ > static const struct snd_kcontrol_new vt1718S_capture_mixer[] = { > HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), > HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, > - HDA_INPUT), > > This would eliminate both boost volume controls at once, but in normal conditions (e.g. front audio panel enabled in bios, headset connected to it) Front Mic Boost works (e.g. I can record sound louder and noisier). I just wondered if creating "Front Mic Boost Capture Volume" could have been done dynamically, instead, to match the driver capability (as is) to avoid creating Volume and Switch controls (on mixer 0x21) for Front Mic: with front audio panel missing (disabled in bios): Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 otherwise: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 A possible solution: /* capture mixer elements */ static const struct snd_kcontrol_new vt1718S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, - HDA_INPUT), ... + static const struct snd_kcontrol_new vt1718S_front_mic_boost = + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT); + static int vt1718S_add_front_mic_boost( struct via_spec *spec ){ + snd_kcontrol_new *knew; + + if( spec->front_panel_out == HDA_HW_WITH_FRONT_PANEL){ + knew = via_clone_control( spec, &vt1718S_front_mic_boost ); + if( knew == NULL ) + return -ENOMEM; + return 1; + } + + return 0; + } in struct via_spec: ... struct hda_pcm pcm_rec[ VIA_NUM_REC]; + #define HDA_HW_WITHOUT_FRONT_PANEL 0 /* initial value*/ + #define HDA_HW_WITH_FRONT_PANEL 1 + unsigned int front_panel_out; /* used to determine the correct number of playback pcm substreams */ in via_new_spec(): ... codec->spec = spec; spec->codec = codec; spec->codec_type = get_codec_type(codec); + spec->front_panel_out = HDA_HW_WITHOUT_FRONT_PANEL; in via_hp_build(): ... knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL){ return -ENOMEM; } knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; + spec->front_panel_out = HDA_HW_WITH_FRONT_PANEL; /* now we know there's a valid secondary playback pcm substream */ in via_build_pcms(): ... info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + /* is this a multistream-capable environment? let's compute real number of playback pcm substreams */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams = 1 + spec->front_panel_out; in patch_vt1718S(): ... if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; + vt1718S_add_front_mic_boost( spec ); } It worked for me, AFAICT: With front audio panel _enabled_ in bios: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Front Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19037: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x7 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1c aplay -l output **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 [...] and front mic boost control listed in amixer: Simple mixer control 'Front Mic Boost',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 3 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] With front audio panel _disabled_ in bios: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x42a190f7: [N/A] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 1 0x1c aplay -l output: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 [...] and control "Front Mic Boost" missing in amixer output. Attaching alsa-info.sh outputs. Perhaps other codecs could benefit of that, like VT1708S, VT1716S, VT2002P, VT1812, all creating a "Front Mic Boost Capture Volume" control statically (each one needing its own modification to vtxxxxx_capture_mixer, patch_vtxxxxx(), and vt1718S_add_front_mic_boost() function renamed, moved upward in the file before patch_vt1708S() and modified with a further snd_kcontrol_new argument to be used by all others, e.g. called as vt_add_front_mic_boost( spec, &vt1718S_front_mic_boost ): + static int vt_add_front_mic_boost( struct via_spec *spec, const struct snd_kcontrol_new *fmb_ctl ){ + snd_kcontrol_new *knew; + + if( spec->front_panel_out == HDA_HW_WITH_FRONT_PANEL){ + knew = via_clone_control( spec, fmb_ctl ); + if( knew == NULL ) + return -ENOMEM; + return 1; + } + + return 0; + } ). > > The problem is in hda_verb vt1718S_volume_init_verbs > > /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, > > node 0x21 of your vt2020 has only 5 connections > > Refer to commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45 > > you will need to ask the author of this patch since the patch > explicitly change this non exisiting connection of node 0x21 on your > vt2020 > > - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, > + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, > > Will do. Perhaps there could be the need to create a separate vt2020_volume_init_verbs, unless that operation is due to internal implementation of the codec (reacting properly to the non-existing connection write query). Thank you for everything. - Nascondi testo citato - > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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Re: [alsa-devel] linux-next: Tree for May 23 (sound/soc/codecs)
by Randy Dunlap 16 Jun '11

16 Jun '11

On Mon, 23 May 2011 15:45:18 +1000 Stephen Rothwell wrote: > Hi all, > > Changes since 20110520: when CONFIG_GPIOLIB is not enabled and CONFIG_GENERIC_GPIO is not enabled: sound/soc/codecs/ak4641.c:524: error: 'GPIOF_OUT_INIT_LOW' undeclared (first use in this function) sound/soc/codecs/wm8915.c:2857: error: 'GPIOF_OUT_INIT_LOW' undeclared (first use in this function) --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] [PATCH] alsa-utils: amixer: Fix SND_CTL_TLVT_DB_RANGE handling
by Lars-Peter Clausen 16 Jun '11

16 Jun '11

Current support for handling SND_CTL_TLVT_DB_RANGE in amixer is completely broken. This patch fixes it. Reported-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- amixer/amixer.c | 14 +++++++------- 1 files changed, 7 insertions(+), 7 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index a177288..9d2855d 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -517,20 +517,20 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #ifdef SND_CTL_TLVT_DB_RANGE case SND_CTL_TLVT_DB_RANGE: printf("dBrange-\n"); - if ((size / (6 * sizeof(unsigned int))) != 0) { + if ((size % (6 * sizeof(unsigned int))) != 0) { while (size > 0) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); } break; } - idx = 0; - while (idx < size) { + while (size > 0) { print_spaces(spaces + 2); - printf("rangemin=%i,", tlv[0]); - printf(",rangemax=%i\n", tlv[1]); - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); - idx += 6 * sizeof(unsigned int); + printf("rangemin=%i,", tlv[idx++]); + printf(",rangemax=%i\n", tlv[idx++]); + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); + idx += 4; + size -= 6 * sizeof(unsigned int); } break; #endif -- 1.7.2.5

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[alsa-devel] [PATCH 0/5] AD1836 updates part two
by Mike Frysinger 16 Jun '11

16 Jun '11

This replaces the patchset I sent out yesterday. It mostly syncs the missing work we had locally with what Lars-Peter did. I guess we can omit the "ad1836" -> "ad183x" rename. Barry Song (1): ASoC: AD1836: rename suspend/resume funcs Mike Frysinger (4): ASoC: AD1836: drop unnecessary spi register check ASoC: AD1836: fix intermixed tab/space indentation ASoC: AD1836: fix codec name ASoC: AD1836: clean up comment headers sound/soc/codecs/ad1836.c | 78 +++++++++++++++++--------------------------- sound/soc/codecs/ad1836.h | 21 +++--------- 2 files changed, 36 insertions(+), 63 deletions(-) -- 1.7.5.3

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add WM8994 PCM Machine driver
by Sangbeom Kim 16 Jun '11

16 Jun '11

This patch add WM8994 PCM machine driver to support PCM audio on SMDKV310, SMDKC210 boards. Playback and Capture supports 8kHz sampling rates. and It is tested on SMDKV310, SMDKC210. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/smdk_wm8994pcm.c | 176 ++++++++++++++++++++++++++++++++++++ 3 files changed, 185 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/smdk_wm8994pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b8c7a2e..db2bc49 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -170,3 +170,11 @@ config SND_SOC_SMDK_WM8580_PCM select SND_SAMSUNG_PCM help Say Y if you want to add support for SoC audio on the SMDK. + +config SND_SOC_SMDK_WM8994_PCM + tristate "SoC PCM Audio support for WM8994 on SMDK" + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + select SND_SOC_WM8994 + select SND_SAMSUNG_PCM + help + Say Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6c598fc..4a3a428 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,6 +35,7 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o +snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -53,3 +54,4 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o +obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c new file mode 100644 index 0000000..756c228 --- /dev/null +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -0,0 +1,176 @@ +/* + * sound/soc/samsung/smdk_wm8994pcm.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "../codecs/wm8994.h" +#include "dma.h" +#include "pcm.h" + +/* + * Board Settings: + * o '1' means 'ON' + * o '0' means 'OFF' + * o 'X' means 'Don't care' + * + * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111 + */ + +/* + * Configure audio route as :- + * $ amixer sset 'DAC1' on,on + * $ amixer sset 'Right Headphone Mux' 'DAC' + * $ amixer sset 'Left Headphone Mux' 'DAC' + * $ amixer sset 'DAC1R Mixer AIF1.1' on + * $ amixer sset 'DAC1L Mixer AIF1.1' on + * $ amixer sset 'IN2L' on + * $ amixer sset 'IN2L PGA IN2LN' on + * $ amixer sset 'MIXINL IN2L' on + * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on + * $ amixer sset 'IN2R' on + * $ amixer sset 'IN2R PGA IN2RN' on + * $ amixer sset 'MIXINR IN2R' on + * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on + */ + +/* SMDK has a 16.9344MHZ crystal attached to WM8994 */ +#define SMDK_WM8994_FREQ 16934400 + +static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned long mclk_freq; + int rfs, ret; + + switch(params_rate(params)) { + case 8000: + rfs = 512; + break; + default: + printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", + __func__, __LINE__, params_rate(params)); + return -EINVAL; + } + + mclk_freq = params_rate(params) * rfs; + + /* Set the codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + SMDK_WM8994_FREQ, mclk_freq); + if (ret < 0) + return ret; + + /* Set PCM source clock on CPU */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set SCLK_DIV for making bclk */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops smdk_wm8994_pcm_ops = { + .hw_params = smdk_wm8994_pcm_hw_params, +}; + +static struct snd_soc_dai_link smdk_dai[] = { + { + .name = "WM8994 PAIF PCM", + .stream_name = "Primary PCM", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .ops = &smdk_wm8994_pcm_ops, + }, +}; + +static struct snd_soc_card smdk_pcm = { + .name = "SMDK-PCM", + .dai_link = smdk_dai, + .num_links = 1, +}; + +static int __devinit snd_smdk_probe(struct platform_device *pdev) +{ + int ret = 0; + + smdk_pcm.dev = &pdev->dev; + ret = snd_soc_register_card(&smdk_pcm); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit snd_smdk_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&smdk_pcm); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_smdk_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "samsung-smdk-pcm", + }, + .probe = snd_smdk_probe, + .remove = __devexit_p(snd_smdk_remove), +}; + +static int __init smdk_audio_init(void) +{ + return platform_driver_register(&snd_smdk_driver); +} + +module_init(smdk_audio_init); + +static void __exit smdk_audio_exit(void) +{ + platform_driver_unregister(&snd_smdk_driver); +} + +module_exit(smdk_audio_exit); + +MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); +MODULE_LICENSE("GPL"); -- 1.7.1

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Re: [alsa-devel] Creative X-Fi Titanium HD ctxfi bugs 5378 5218 4868
by Takashi Iwai 16 Jun '11

16 Jun '11

At Sat, 11 Jun 2011 18:25:36 +0800, Harry Butterworth wrote: > > Hi, > > I have redone the patch against today's git repositories. > Creative have not yet got back to me about the PLL initialization. > Perhaps we can fix that later. > > I have only put the PCI ID in the pci_ids_compat.h.in file. Is it > going to be necessary to add the PCI device ID somewhere in the kernel > tree to avoid breaking the kernel build? Yes, include/linux/pci_ids.h is the place to change. But I can fix it by myself, too. > > > A few comment about the patch: > > > > - Instead of adding have_dedicated_mic and have_output_switch > > callbacks in every place, I'd put some bit flags indicating the > > capabilities, and set at the initialization appropriately. > > I had a go at this in the patch 0002-ALSA... > > > - A few easy coding-style issues; run scripts/checkpatch.pl and fix > > the errors appropriately. > > checkpatch now reports no errors. > > I've only tested these changes with my Titanium HD card. I don't have > any of the other X-Fi cards so I couldn't do any regression testing. > > If there's anything else I need to do, please let me know. I'm going to check patches in a more detail tomorrow. Thanks! Takashi

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[alsa-devel] [PATCH] udev: Allow ALSA input jacks to be accessed by the current user
by David Henningsson 15 Jun '11

15 Jun '11

I know this is probably not the correct upstream, but I just wanted to run it through alsa-devel first to make sure that you don't think this is all madness. :-) So, one missing piece for userspace (PulseAudio etc) to actually be able to use the jack input devices that ALSA create, is that these devices are accessible by root only. This patch makes the input device nodes accessible by the same users that can access the sound card: the current logged in user, as well as users in the audio group. One thing I was thinking about, was that the udev-acl rule actually grants read-write access to the input device node, where probably only read access is needed. Is this dangerous? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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