May 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/2] ASoC: TWL4030: Add voice digital loopback: sidetone
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++-- 1 files changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efb371f..23bae74 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -491,6 +491,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = TWL4030_REG_ATX2ARXPGA, 0, 7, 0, twl4030_dapm_dbypass_tlv); +/* + * Voice Sidetone GAIN volume control: + * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB) + */ +static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1); + +/* Digital bypass voice: sidetone (VUL -> VDL)*/ +static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_VSTPGA, 0, 0x29, 0, + twl4030_dapm_dbypassv_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -607,12 +619,18 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); + } else if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); + twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); + twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); } /* Enable master analog loopback mode if any analog switch is enabled*/ @@ -991,6 +1009,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control, bypass_event, + SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), @@ -1203,9 +1224,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- 1.5.4.3

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[alsa-devel] Issues with HP DV4 laptop sound after Ubuntu Jaunty 9.04 upgrade
by Guy Stalnaker 03 May '09

03 May '09

Hello again. I was able to get my internal speakers working for my HP Pavilion Dv4 laptop by following the advice I found in the alsa-devel archives. That advice seems to have worked for quit a few people who found it by a post I made to the Ubuntu Forums. But now, after I have upgraded my Ubuntu from version 8.10 (when sound worked) to version 9.04, sound no longer works correctly. I have some detailed information about the sequencing and the specific error. Attached is an alsa-info.txt output. Here's what I'm seeing (extracts from dmesg/kernel logs): Apr 29 15:13:17 Harlindon kernel: [ 19.937190] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 29 15:13:17 Harlindon kernel: [ 19.937270] HDA Intel 0000:00:14.2: setting latency timer to 64 Apr 30 09:37:09 Harlindon kernel: [ 19.941675] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 30 09:37:09 Harlindon kernel: [ 19.941750] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:06:01 Harlindon kernel: [ 19.852324] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:06:01 Harlindon kernel: [ 19.852398] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:13:43 Harlindon kernel: [ 20.950985] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:13:43 Harlindon kernel: [ 20.951060] HDA Intel 0000:00:14.2: setting latency timer to 64 Commit Log for Fri May 1 16:41:58 2009 [Upgrade to Ubuntu Jaunty 9.04] May 1 17:11:05 Harlindon kernel: [ 20.318416] hda-intel: unable to grab IRQ 0, disabling device May 1 17:11:05 Harlindon kernel: [ 20.318517] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:14:18 Harlindon kernel: [ 22.977762] hda-intel: unable to grab IRQ 0, disabling device May 1 17:14:18 Harlindon kernel: [ 22.977828] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:18:49 Harlindon kernel: [ 21.105013] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 17:18:49 Harlindon kernel: [ 21.105087] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 21:15:14 Harlindon kernel: [ 19.776991] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 21:15:14 Harlindon kernel: [ 19.777893] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:22:01 Harlindon kernel: [ 20.540543] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 09:22:01 Harlindon kernel: [ 20.540619] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:26:04 Harlindon kernel: [ 19.819578] hda-intel: unable to grab IRQ 0, disabling device May 2 09:26:04 Harlindon kernel: [ 19.819677] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 10:37:28 Harlindon kernel: [ 20.175934] hda-intel: unable to grab IRQ 0, disabling device May 2 10:37:28 Harlindon kernel: [ 20.175999] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 13:26:27 Harlindon kernel: [ 20.610591] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 13:26:27 Harlindon kernel: [ 20.610666] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 16:57:23 Harlindon kernel: [ 19.885497] hda-intel: unable to grab IRQ 0, disabling device May 2 16:57:23 Harlindon kernel: [ 19.885575] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 18:20:18 Harlindon kernel: [ 22.605977] hda-intel: unable to grab IRQ 0, disabling device May 2 18:20:18 Harlindon kernel: [ 22.606042] HDA Intel: probe of 0000:00:14.2 failed with error -16 What is damned peculiar is that is has sometimes worked (as you can see above). I can find no pattern however. I HAVE downloaded an compiled the latest alsa-driver-snapshot. That last reboot has the same unable to grab irq error. Any suggestions are greatly appreciated. -- Guy Stalnaker, I^2@DOIT, 1210 West Dayton Street, Room 3209 CSS, Madison WI 53719-1220, jstalnak(a)wisc.edu, work 608.263.8035, cell 608.235.4718, fax 608.265.6681, page page-guy(a)watchdog.doit.wisc.edu upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Sun May 3 04:38:21 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!Kernel Information !!------------------ Kernel release: 2.6.28-6-386 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: No !!ALSA Version !!------------ Driver version: 1.0.19 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xd2410000 irq 19 !!PCI Soundcards installed in the system !!-------------------------------------- 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:14.2 0403: 1002:4383 Subsystem: 103c:30fb -- 01:05.1 0403: 1002:960f Subsystem: 1002:960f !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd_hda_intel: model=hp-dv5 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: ATI RS690/780 HDMI Address: 0 Function Id: 0x1 Vendor Id: 0x1002791a Subsystem Id: 0x00791a00 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x40]: 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 May 2 18:25 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 May 2 18:25 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 4 May 2 18:25 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 3 May 2 18:25 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 May 2 18:25 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** !!Amixer output !!------------- !!-------Mixer controls for card 0 [HDMI] Card hw:0 'HDMI'/'HDA ATI HDMI at 0xd2410000 irq 19' Mixer name : 'ATI RS690/780 HDMI' Components : 'HDA:1002791a,00791a00,00100000' Controls : 4 Simple ctrls : 1 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.HDMI { control.1 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.2 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.3 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module ppp_deflate zlib_deflate bsd_comp ppp_async crc_ccitt ppp_generic slhc binfmt_misc ppdev rfcomm bridge stp bnep sco l2cap bluetooth ipv6 container sbs pci_slot sbshc rfkill input_polldev cpufreq_powersave cpufreq_stats cpufreq_userspace cpufreq_ondemand cpufreq_conservative freq_table cbc dm_crypt sha256_generic sha1_generic sha512_generic aes_i586 aes_generic lp parport joydev snd_hda_codec_atihdmi snd_hda_intel snd_seq_dummy snd_hda_codec snd_seq_oss snd_seq_midi snd_hwdep snd_rawmidi snd_pcm_oss snd_mixer_oss snd_seq_midi_event psmouse i2c_piix4 snd_pcm uvcvideo compat_ioctl32 serio_raw snd_seq i2c_core sierra snd_timer snd_seq_device videodev v4l1_compat usblp evdev usbserial fglrx snd soundcore snd_page_alloc agpgart battery leds_hp_disk led_class shpchp pci_hotplug video output lis3lv02d sdhci_pci sdhci mmc_core ac button reiserfs ata_generic pata_acpi pata_atiixp sg sr_mod cdrom sd_mod crc_t10dif usbhid hid atiixp ahci libata r8169 mii usb_storage scsi_mod dm_mirror dm_region_hash dm_log dm_snapshot dm_mod thermal processor fan fbcon tileblit font bitblit softcursor !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x03 0x18560010 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs:

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Re: [alsa-devel] [Alsa-user] looking for advice on buying a card
by Dmitri Seletski 02 May '09

02 May '09

Hi Gary. Well, thank you for the idea, but i somehow not comfortable with it. I have to buy this thing, even if it's inexpensive, i will have to pay good amount of money for delivery. I am looking around 20 euro, not counting price for device itself. Than again, i have to buy new card and pay for delivery there. To make things worse, none xfi sound cards are no longer produced by creative labs and i really don't like idea of second hand sound cards(even though you have to be very skillful to break one of those). So it simply makes sense to pay a bit extra(i was loosing those money with that pcie to pci extentsion and its delivery anyway) but to get that card I was referring before, Asus Xonar DX I know it's supported by alsa, because of this: http://www.alsa-project.org/main/index.php/Matrix:Vendor-Asus I am just double checking, since this specific card is really promising one and i really feel like buying one. But again, i am opened for other options. Thanks a lot for your thoughts. I have kinda got used to USB cards by now myself and all those troubles they cause :) gary wrote: > Having spent an ungodly obscene amount of time trying to get two > different USB audio cards to work in 64 bit opensuse, I suggest not > trying to be a pioneer. I solved my need for multiple sound cards > (exceeding the number of PCI slots I have) by using a PCI bridge. > > For one soundcard, you can try thing: > http://www.siig.com/ViewProduct.aspx?pn=JJ-000111-S1 > It doesn't list linux in support unfortunately. However, the latest > linux kernal might recognize the card. Then pick a soundcard from the > list of supported devices. > > For my particular situation, I bought a Magma PCI expander on ebay. I > plugged it in and it worked with the soundcards I had on hand (Diamond > 5.1 using C-Media chipset). With the shipping, it worked out to be > about $39 a PCI slot. [My time, about 10 minutes, not counting > searching ebay.] Thus the price for the SIIG seems reasonable if you > only need to run one card. However, I can't vouch it will work. The > Magma boxes do work, but are obscenely expensive new and bulky. > > > > Dmitri Seletski wrote: >> Hello Guys >> I am thinking to buy pci express sound card. Nothing fancy, just >> something with dolby surround 7.1 and decent sound quality, rather >> inexpensive, i am not audiophile. >> >> I have noticed that there are not that many sound cards out there which >> have descent support in Linux(unlike many forums claim). I have had >> MAJOR problems with pci xfi card from creative labs(fatality gamers >> edition), my buddy still has a lot of problem with it, like major issues >> with input. No mic - no skype or other voip, you know yourself... >> >> I have found source of asus xonar dx(namely ebay), it is not that >> expensive compared to many other pci express cards out there, still is >> expensive enough, from alsa matrix it appears to be quite descent. >> Question is, is it worth buying? Is it painful to get it >> running?(probably nothing unusual, matter of loading module), does it >> work fully(does microphone work, does surround sound 7.1 work?)? >> Is there something better for similar money?(EUR 73 or £65 or $97) >> Please do not propose incompletely supported cards(while i wouldn't mind >> submitting issues to mail lists if i come across bugs, i wouldn't want >> product that doesn't work) >> >> I appreciate any comments. >> >> Thank you in advance. >> Dmitri >> >> ------------------------------------------------------------------------------ >> >> Register Now & Save for Velocity, the Web Performance & Operations >> Conference from O'Reilly Media. Velocity features a full day of >> expert-led, hands-on workshops and two days of sessions from industry >> leaders in dedicated Performance & Operations tracks. Use code >> vel09scf and Save an extra 15% before 5/3. >> http://p.sf.net/sfu/velocityconf >> _______________________________________________ >> Alsa-user mailing list >> Alsa-user(a)lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/alsa-user >> >> >> ------------------------------------------------------------------------ >> >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: >> 270.12.13/2091 - Release Date: 05/01/09 17:52:00 >> > ------------------------------------------------------------------------ > > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.15/2093 - Release Date: 05/02/09 14:23:00 > >

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[alsa-devel] New codec for STAC9766 used on Efika
by Jon Smirl 02 May '09

02 May '09

Could you please review this new STAC9766 codec implementation. Some of the controls aren't where they need to be. For example the mutes on the mixer inputs. I'm not sure where they are supposed to go. How should SPDIF support work? The hardware supports simultaneous use of analog and SPDIF. Do I want one DAI or two? If I have two DAI then ALSA needs to interleave the samples into a single DMA stream. So if one stream isn't active it needs to be filled with silence. Other options... A single DAI with a switch A single DAI with the same output stream on both analog and SPDIF. I've tried to add the DAPM routes. I don't know of anyone using this chip on a low power design. Datasheet is here: http://www.idt.com/products/getDoc.cfm?docID=13134007 I get a few errors from the current config: Failed to add route All Analog Mixer->Pop Bypass Mux dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack /* * stac9766.c -- ALSA Soc STAC9766 codec support * * Copyright 2008 Jon Smirl, Digispeaker * Author: Jon Smirl <jonsmirl(a)gmail.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. * * Features:- * * o Support for AC97 Codec, S/PDIF * o Support for DAPM */ #include <linux/init.h> #include <linux/module.h> #include <linux/device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-of-simple.h> #include "stac9766.h" #define STAC9766_VERSION "0.10" /* * STAC9766 register cache */ static const u16 stac9766_reg[] = { 0x6A90, 0x8000, 0x8000, 0x8000, // 6 0x0000, 0x0000, 0x8008, 0x8008, // e 0x8808, 0x8808, 0x8808, 0x8808, // 16 0x8808, 0x0000, 0x8000, 0x0000, // 1e 0x0000, 0x0000, 0x0000, 0x000f, // 26 0x0a05, 0x0400, 0xbb80, 0x0000, // 2e 0x0000, 0xbb80, 0x0000, 0x0000, // 36 0x0000, 0x2000, 0x0000, 0x0100, // 3e 0x0000, 0x0000, 0x0080, 0x0000, // 46 0x0000, 0x0000, 0x0003, 0xffff, // 4e 0x0000, 0x0000, 0x0000, 0x0000, // 56 0x4000, 0x0000, 0x0000, 0x0000, // 5e 0x1201, 0xFFFF, 0xFFFF, 0x0000, // 66 0x0000, 0x0000, 0x0000, 0x0000, // 6e 0x0000, 0x0000, 0x0000, 0x0006, // 76 0x0000, 0x0000, 0x0000, 0x0000, // 7e }; static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; static const char *stac9766_3D_separation[] = {"Off", "Low", "Medium", "High"}; static const struct soc_enum stac9766_enum[] = { SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux), /* Record Mux 0 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux), /* Mono Mux 1 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux), /* Mic1/2 Mux 2 */ SOC_ENUM_SINGLE(AC97_SENSE_INFO, 1, 2, stac9766_SPDIF_mux), /* SPDIF Mux 3 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux), /* Pop Bypass Mux 4 */ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux), /* Record All Mux 5 */ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 2, 4, stac9766_3D_separation), /* 3D Separation 7 */ }; static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Mixer PC Beep Volume", AC97_PC_BEEP, 1, 15, 1), SOC_SINGLE("PC Beep Frequency", AC97_PC_BEEP, 5, 127, 1), SOC_SINGLE("Mixer Phone Volume", AC97_PHONE, 0, 31, 1), SOC_SINGLE("Mixer Mic Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic Boost", AC97_MIC, 6, 1, 1), SOC_SINGLE("Mic Gain", AC97_STAC_ANALOG_SPECIAL, 2, 1, 1), SOC_SINGLE("Stereo Mic", AC97_STAC_STEREO_MIC, 0, 1, 1), SOC_DOUBLE("Mixer Line Volume", AC97_LINE, 8, 0, 31, 1), SOC_DOUBLE("Mixer CD Volume", AC97_CD, 8, 0, 31, 1), SOC_DOUBLE("Mixer Video Volume", AC97_VIDEO, 8, 0, 31, 1), SOC_DOUBLE("Mixer AUX Volume", AC97_AUX, 8, 0, 31, 1), SOC_DOUBLE("All Analog PCM Volume", AC97_PCM, 8, 0, 31, 1), SOC_DOUBLE("Record Gain", AC97_REC_GAIN, 8, 0, 31, 1), SOC_SINGLE("Record Gain Switch", AC97_REC_GAIN, 15, 1, 1), SOC_SINGLE("3D Effect Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), SOC_ENUM("3D Separation", stac9766_enum[6]), }; /* Mixer */ static const struct snd_kcontrol_new stac9766_main_mixer_controls[] = { SOC_DAPM_SINGLE("PC Beep Switch", AC97_PC_BEEP, 15, 1, 1), SOC_DAPM_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), SOC_DAPM_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), SOC_DAPM_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), SOC_DAPM_SINGLE("CD Switch", AC97_CD, 15, 1, 1), SOC_DAPM_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), SOC_DAPM_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), }; /* Record All Mixer */ static const struct snd_kcontrol_new stac9766_record_all_mixer_controls[] = { SOC_DAPM_SINGLE("PCM Switch", AC97_PCM, 15, 1, 1), SOC_DAPM_SINGLE("Mixer", 0, 0, 0, 0), }; /* Record Mux 0 */ static const struct snd_kcontrol_new stac9766_record_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[0]); /* Mono Mux 1 */ static const struct snd_kcontrol_new stac9766_mono_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[1]); /* Mic1/2 Mux 2 */ static const struct snd_kcontrol_new stac9766_mic_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[2]); /* SPDIF Mux 3 */ static const struct snd_kcontrol_new stac9766_spdif_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[3]); /* Pop Bypass 4 */ static const struct snd_kcontrol_new stac9766_popbypass_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[4]); /* Record All 5 */ static const struct snd_kcontrol_new stac9766_record_all_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[5]); static const struct snd_soc_dapm_widget stac9766_dapm_widgets[] = { SND_SOC_DAPM_INPUT("PCMOut"), SND_SOC_DAPM_INPUT("PCBEEP"), SND_SOC_DAPM_INPUT("Phone"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("Line"), SND_SOC_DAPM_INPUT("CD"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_INPUT("Video"), SND_SOC_DAPM_DAC("DAC", "Analog Playback", AC97_POWERDOWN, 9, 1), SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, &stac9766_spdif_mux_controls), SND_SOC_DAPM_MUX("Mic1/2 Mux", SND_SOC_NOPM, 0, 0, &stac9766_mic_mux_controls), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 1, &stac9766_main_mixer_controls[0], ARRAY_SIZE(stac9766_main_mixer_controls)), SND_SOC_DAPM_MUX("Record All Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_all_mux_controls), SND_SOC_DAPM_MUX("Record Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_mux_controls), SND_SOC_DAPM_MUX("Mono Mux", SND_SOC_NOPM, 0, 0, &stac9766_mono_mux_controls), SND_SOC_DAPM_MUX("Pop Bypass Mux", SND_SOC_NOPM, 0, 0, &stac9766_popbypass_mux_controls), SND_SOC_DAPM_MIXER("All Analog", SND_SOC_NOPM, 0, 1, &stac9766_record_all_mixer_controls[0], ARRAY_SIZE(stac9766_record_all_mixer_controls)), SND_SOC_DAPM_ADC("ADC", "Analog Capture", AC97_POWERDOWN, 8, 1), SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_OUTPUT("LINE"), SND_SOC_DAPM_OUTPUT("MONO"), SND_SOC_DAPM_OUTPUT("PCMIn"), SND_SOC_DAPM_VMID("VMID"), }; static const struct snd_soc_dapm_route audio_map[] = { {"HP", NULL, "Pop Bypass Mux"}, {"LINE", NULL, "Pop Bypass Mux"}, /* Mono Mux */ {"MONO", NULL, "Pop Bypass Mux"}, {"MONO", NULL, "Mic1/2 Mux"}, /* Pop Bypass Mux */ {"Pop Bypass Mux", NULL, "DAC"}, {"Pop Bypass Mux", NULL, "All Analog Mixer"}, /* Record Mux */ {"ADC", NULL, "Mic1/2 Mux"}, {"ADC", NULL, "CD"}, {"ADC", NULL, "Video"}, {"ADC", NULL, "AUX"}, {"ADC", NULL, "Line"}, {"ADC", NULL, "Record All Mux"}, {"ADC", NULL, "Phone"}, {"PCMIn", NULL, "ADC"}, /* Record All Mux */ {"Record All Mux", NULL, "Mixer"}, {"Record All Mux", NULL, "All Analog Mixer"}, /* All Analog Mixer */ {"All Analog", NULL, "Mixer"}, {"All Analog", "PCM Switch", "DAC"}, /* Mixer */ {"Mixer", "PC Beep Switch", "PCBEEP"}, {"Mixer", "Phone Switch", "Phone"}, {"Mixer", "Mic Switch", "Mic1/2 Mux"}, {"Mixer", "Line Switch", "Line"}, {"Mixer", "CD Switch", "CD"}, {"Mixer", "AUX Switch", "AUX"}, {"Mixer", "Video Switch", "Video"}, {"DAC", NULL, "PCMOut"}, /* Mic1/2 Mux */ {"Mic1/2 Mux", NULL, "MIC1"}, {"Mic1/2 Mux", NULL, "MIC2"}, /* SPDIF Mux */ {"SPDIF Mux", NULL, "PCMOut"}, {"SPDIF Mux", NULL, "ADC"}, {NULL, NULL, NULL}, }; static int stac9766_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, stac9766_dapm_widgets, ARRAY_SIZE(stac9766_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; } unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { val = soc_ac97_ops.read(codec->ac97, reg); return val; } return cache[reg / 2]; } int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { u16 *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } static int ac97_analog_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; unsigned short reg, vra; vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); //vra |= 0x4; stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); printk("AC97_EXTENDED_STATUS %x\n", vra); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; return stac9766_ac97_write(codec, reg, runtime->rate); } static int ac97_digital_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { printk("stac9766: ac97_digital_prepare\n"); return 0; } static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_PREPARE: /* partial On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_STANDBY: /* Off, with power */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* Off, without power */ /* disable everything including AC link */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } codec->bias_level = level; return 0; } static int stac9766_codec_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int stac9766_codec_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; u16 id; /* give the codec an AC97 warm reset to start the link */ codec->ac97->bus->ops->warm_reset(codec->ac97); id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { printk(KERN_ERR "stac9766 failed to resume"); return -EIO; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (codec->suspend_bias_level == SND_SOC_BIAS_ON) stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); return 0; } static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, }; struct snd_soc_dai stac9766_dai[] = { { .name = "stac9766 analog", .id = 0, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, /* alsa ops */ .ops = &stac9766_dai_ops_analog, }, { .name = "stac9766 digital", .id = 1, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 digital", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME, }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, }}; EXPORT_SYMBOL_GPL(stac9766_dai); int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); if (soc_ac97_ops.warm_reset) soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; } static int stac9766_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; int ret = 0; printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) return -ENOMEM; codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; } codec->reg_cache_size = sizeof(stac9766_reg); codec->reg_cache_step = 2; codec->name = "STAC9766"; codec->owner = THIS_MODULE; codec->dai = stac9766_dai; codec->num_dai = ARRAY_SIZE(stac9766_dai); codec->write = stac9766_ac97_write; codec->read = stac9766_ac97_read; codec->set_bias_level = stac9766_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) goto pcm_err; /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ stac9766_reset(codec, 0); ret = stac9766_reset(codec, 1); if (ret < 0) { printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); goto reset_err; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); stac9766_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) goto reset_err; return 0; reset_err: snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: kfree(codec->private_data); cache_err: kfree(socdev->card->codec); socdev->card->codec = NULL; return ret; } static int stac9766_codec_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; snd_soc_dapm_free(socdev); snd_soc_free_pcms(socdev); snd_soc_free_ac97_codec(codec); kfree(codec->reg_cache); kfree(codec); return 0; } struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); static int __init stac9766_probe(struct platform_device *pdev) { #if defined(CONFIG_SND_SOC_OF_SIMPLE) /* Tell the of_soc helper about this codec */ of_snd_soc_register_codec(&soc_codec_dev_stac9766, pdev->dev.archdata.of_node, stac9766_dai, ARRAY_SIZE(stac9766_dai), pdev->dev.archdata.of_node); #endif return 0; } static struct platform_driver stac9766_driver = { .probe = stac9766_probe, .driver = { .name = "stac9766", }, }; static __init int stac9766_driver_init(void) { snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); return platform_driver_register(&stac9766_driver); } static __exit void stac9766_driver_exit(void) { } module_init(stac9766_driver_init); module_exit(stac9766_driver_exit); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); MODULE_LICENSE("GPL"); -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] errors in wm8990 codec driver
by Park Jinyoung 01 May '09

01 May '09

Hello, I made wm8990 patch file on your request. The patch file was attached in this. I use my company email, because I have no other method for file attach in my company. Thanks. 2009/4/30 Mark Brown <broonie(a)sirena.org.uk>: > On Thu, Apr 30, 2009 at 02:03:03PM +0900, Jinyoung Park wrote: >> I found a number of typing errors in wm8990 codec driver. >> The mis-typing exist in dapm controller definitions and dapm route definitions, >> so happen mis-matched error when snd_soc_dapm_add_routes(). >> Belows are my modified code. Please check... > > This all looks reasonable - could you please supply a Signed-off-by for > your patch? It may also help to use git send-email to send your patch > or attach it, gmail tends to mangle patches so that they can't be > applied and it looks like some of that may have happened here, though > I've not actually tried applying it. >

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