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[alsa-devel] [PATCH] USB quirk for Logitech Quickcam Pro 9000 name
by Peter Stokes 14 Jan '09

14 Jan '09

Hi, The Logitech QuickCam Pro 9000 does not appear to any product identification strings in its USB device descriptor. Therefore it receives a device name of "USB Device 0x46d:0x990". The attached patch below adds a USB quirk to provide a more friendly name. Also, for your reference, please find attached the output from "lsusb -vv -d 046d:0990" Best regards Peter Stokes ------------------------------------------------------------------------------- Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk> diff --git a/usb/usbquirks.h b/usb/usbquirks.h index 9211575..5d8ef09 100644 --- a/usb/usbquirks.h +++ b/usb/usbquirks.h @@ -128,6 +128,14 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + USB_DEVICE(0x046d, 0x0990), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Logitech, Inc.", + .product_name = "QuickCam Pro 9000", + .ifnum = QUIRK_NO_INTERFACE + } +}, /* * Yamaha devices

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[alsa-devel] [PATCH] hda: stac92hd8xxx amp mixers
by Matthew Ranostay 14 Jan '09

14 Jan '09

Added amp nid for stac92hd8xxx families of codecs so the input amp mixer is created. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 35b83dc..dedf8b2 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -331,6 +331,10 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x10, 0x40, }; +static unsigned int stac92hd83xxx_amp_nids[1] = { + 0xc, +}; + static hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; @@ -4672,6 +4676,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->dmux_nids = stac92hd83xxx_dmux_nids; spec->adc_nids = stac92hd83xxx_adc_nids; spec->pwr_nids = stac92hd83xxx_pwr_nids; + spec->amp_nids = stac92hd83xxx_amp_nids; spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; @@ -4689,6 +4694,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); + spec->num_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids); spec->num_dmics = STAC92HD83XXX_NUM_DMICS; spec->dinput_mux = &stac92hd83xxx_dmux; spec->pin_nids = stac92hd83xxx_pin_nids;

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Re: [alsa-devel] [PATCH] ALSA: hda - Add automatic model setting for Samsung Q45
by Luke Yelavich 14 Jan '09

14 Jan '09

Sorry to take so long to reply. This quirk was from Ubuntu bug 200210, http://launchpad.net/bugs/200210. The patch was based on a comment from Daniel T Chen, indicating the quirk needed for the bug. Luke

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[alsa-devel] set_params:1073: not enough memory
by Michael Hirsch 13 Jan '09

13 Jan '09

Hello all, I'm trying to get alsa 1.0.18 working on a ARM PXA processor and I'm stuck with the following error message: aplay /Sinus.raw -fcd Playing raw data '/Sinus.raw' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1073: not enough memory alsa kernel was compiled with alsa 1.0.16, alsa.conf, located in /usr/share/alsa is the basic: pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } also aplay -l and alsaconf store show the message: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 However the card shows up /proc/asound/cards and alsamixer is able to access it. Any ideas? Best regards Michael

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Re: [alsa-devel] Fixing the AICA driver
by Takashi Iwai 13 Jan '09

13 Jan '09

At Thu, 08 Jan 2009 18:50:46 +0000, Adrian McMenamin wrote: > > On Wed, 2009-01-07 at 12:02 +0100, Takashi Iwai wrote: > > > > > Could you try the git version? I thought I fixed the relevant bug recently. > > > > > > Takashi > > Hi Takashi - I assume you mean in the alssa-driver git, as I am using > the bleeding edge kernel stuff already? No, I mean alsa-lib. There is mmap_emul plugin in alsa-lib but it didn't work well. Takashi

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[alsa-devel] Recorded frames sign 'error' after switchign to EABI kernel?
by Edwin van der Horst 13 Jan '09

13 Jan '09

Hello all, I'm using the ALSA API for quite some time now in my application, all worked fine until I switched to EABI kernel/compiler. My application reads signed 16 bit frames (S16_LE) using readi call: long FCAudio::readRecordedPCM(char * dest,int len) { .... frames = snd_pcm_readi(recHandle, dest, (len / asRec.bytesPS)); //read data hw .... .... } This worked fine on the non EABi kernel. After I switched to the EABI enabled kernel and compiler, I heard massive 'hiss/noise' on recorded audio. I took a dump of the recorded frames, and this came out: sample data: | ff | 58 | ff | e1 | ff | e2 | 0 | 0 | 0 | 48 | ff | e0 | ff | e3 | ff | e1 | 0 | 13 | ff | ca | ff | ae | 0 | 3 | ff | ee | ff | be| at first this seemed like correct data to me , but on a closer look the MSB(yte) is always 0x00 or 0xff I can't figure out why this is: I'm expecting 2 bytes of sample data, and in the current case I get 1 byte of correct data, and the other byte is only used for sign indication (0x00 or 0xFF). I can't figure out why this is, can anyone point me in the right direction? It's worth noting that arecord and aplay both work correctly, and using snd_pcm_dump(); I get settings as expedted and are correct. Thanks in advance... Edwin van der Horst

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[alsa-devel] [PATCH - SB0500 2/2] rename "Device" to "Toshiba SB-0500" via quirks
by Andrea Borgia 13 Jan '09

13 Jan '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/usb/usbquirks.h b/usb/usbquirks.h index 9211575..d59323e 100644 --- a/usb/usbquirks.h +++ b/usb/usbquirks.h @@ -39,6 +39,16 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* Creative/Toshiba Multimedia Center SB-0500 */ +{ + USB_DEVICE(0x041e, 0x3048), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Toshiba", + .product_name = "SB-0500", + .ifnum = QUIRK_NO_INTERFACE + } +}, + /* Creative/E-Mu devices */ { USB_DEVICE(0x041e, 0x3010), -- 1.5.4.3

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[alsa-devel] [PATCH - SB-0500 1/1] preliminary support for Toshiba SB-0500
by Andrea Borgia 13 Jan '09

13 Jan '09

The Toshiba Multimedia Center SB-0500 is a rebranded version of the Creative Technology SB Live! 24-bit External: it shares the same chipset and only has minor cosmetic differences. Remote controller works with alsa_usb module, basic audio is there and mixer controls are mostly untested. Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> --- diff --git a/usb/usbmixer.c b/usb/usbmixer.c index a492461..6490a74 100644 --- a/usb/usbmixer.c +++ b/usb/usbmixer.c @@ -66,6 +66,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; struct usb_mixer_interface { @@ -1706,7 +1707,8 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, break; /* live24ext: 4 = line-in jack */ case 3: /* hp-out jack (may actuate Mute) */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); break; default: @@ -1957,8 +1959,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { - if (i > 1 && /* Live24ext has 2 LEDs only */ - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (i > 1 && /* Live24ext has 2 LEDs only */ + (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); @@ -1995,7 +1998,8 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) jacks = jacks_audigy2nx; - else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) jacks = jacks_live24ext; else return; @@ -2045,7 +2049,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, goto _error; if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) { + mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { struct snd_info_entry *entry; if ((err = snd_audigy2nx_controls_create(mixer)) < 0) diff --git a/usb/usbmixer_maps.c b/usb/usbmixer_maps.c index d755be0..f41214f 100644 --- a/usb/usbmixer_maps.c +++ b/usb/usbmixer_maps.c @@ -285,6 +285,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = live24ext_map, }, { + .id = USB_ID(0x041e, 0x3048), + .map = audigy2nx_map, + .selector_map = audigy2nx_selectors, + }, + { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), .ignore_ctl_error = 1, -- 1.5.4.3 !DSPAM:49626d6b49731845111285!

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[alsa-devel] 'node->type == SND_CONFIG_TYPE_COMPOUND' failed
by Michael Hirsch 12 Jan '09

12 Jan '09

Hello all, I'm trying to get alsa 1.0.16 working on a ARM PXA processor and I'm stuck with the following error message: aplay: conf.c: 3166: snd_config_iterator_first: Assertion `node->type == SND_CONFIG_TYPE_COMPOUND' failed. kernel version matches the user space library and utils version. searching the archives / google brings no result. We tried with an empty alsa.conf but then the message: ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:564: audio open error: No such file or directory appears. However the card shows up /proc/asound/cards and alsamixer is able to access it. Any ideas? Best regards Michael

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Re: [alsa-devel] [PATCH] wss-lib: move AD1845 frequency setting into wss-lib
by Krzysztof Helt 12 Jan '09

12 Jan '09

On Sun, 11 Jan 2009 08:28:13 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > The Soundscape Elite does work, but after a reboot > I noticed that there is an oddness in that there's > a little "pop" before playing sounds with aplay and > a series of short spike pops before the sound plays > with xmms, which uses the old oss interface. > I traced this and the "pop" sound is caused by two things: 1. Codec autocalibration every time a new sound is started (even when sound is muted it produces the "pop" on AD1848). 2. Muting sound on and off every autocalibration. The OSS code from the 2.4 kernel states that autocalibration produced that "pop" so it got disabled. This solves two problems at once as the muting on and off is not required then (and is not done in the code). I am not audiofile so I cannot comment how audible the lack of autocalibration is. Most codecs require it after they are powered up (and most of them except the oldest ones actually force it after powering up). One can autocalibrate the codec only once during a card/codec initialization. I can prepare a patch which disables the autocalibration. This is the only solution as I tested others like slow volume muting but all others still produce some noise. Regards, Krzysztof ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

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[alsa-devel] snd_usb_caiaq bug?
by Marco Santamaria 12 Jan '09

12 Jan '09

Hi, it's about one year that I'm using my Native Instruments Audio Kontrol 1 (USB2 audio/MIDI interface) only with audio applications and it has been working well so far. Now I'm trying to use it with Rosegarden (MIDI) and, as soon as I start Rosegarden, JACK crashes, the process jackd becomes "uninterruptible" and with the command dmesg I find the following new lines [ 1935.173876] BUG: unable to handle kernel NULL pointer dereference at virtual address 00000044 [ 1935.173885] printing eip: f8a5f6fd *pde = 00000000 [ 1935.173891] Oops: 0000 [#1] PREEMPT SMP [ 1935.173896] Modules linked in: isofs udf ipv6 af_packet i915 drm rfcomm l2cap bluetooth ppdev acpi_cpufreq cpufreq_ondemand cpufreq_conservative cpufreq_stats freq_table cpufreq_userspace cpufreq_powersave dock container sbs sbshc iptable_filter ip_tables x_tables sbp2 parport_pc lp parport snd_hda_intel snd_usb_caiaq snd_hwdep snd_pcm_oss snd_mixer_oss arc4 snd_seq_oss snd_pcm ecb blkcipher snd_seq_midi snd_page_alloc joydev snd_rawmidi snd_seq_midi_event snd_seq iwl3945 sdhci serio_raw mmc_core iwlwifi_mac80211 video output cfg80211 battery led_class ac snd_timer snd_seq_device wmi_acer button snd evdev dcdbas iTCO_wdt iTCO_vendor_support intel_agp shpchp pci_hotplug pcspkr soundcore agpgart psmouse ext3 jbd mbcache sg sr_mod cdrom sd_mod b44 ata_generic ssb mii ohci1394 ieee1394 ata_piix pata_acpi libata scsi_mod ehci_hcd uhci_hcd usbcore thermal processor fan fuse [ 1935.173964] [ 1935.173967] Pid: 1421, comm: IRQ-18 Not tainted (2.6.24-19-rt #1) [ 1935.173970] EIP: 0060:[<f8a5f6fd>] EFLAGS: 00010292 CPU: 1 [ 1935.173979] EIP is at snd_rawmidi_transmit_peek+0x1d/0x100 [snd_rawmidi] [ 1935.173982] EAX: 00000000 EBX: f7106264 ECX: 0000003d EDX: f7106407 [ 1935.173985] ESI: ffffffea EDI: 00000000 EBP: f7106407 ESP: df85be38 [ 1935.173988] DS: 007b ES: 007b FS: 00d8 GS: 0000 SS: 0068 preempt:00000001 [ 1935.173991] Process IRQ-18 (pid: 1421, ti=df85a000 task=f7d45320 task.ti=df85a000) [ 1935.173993] Stack: 0000003d 00000000 f7106264 00000000 00000000 df83fc00 f8b8cc09 00000000 [ 1935.174000] f7106264 f710630c 00000000 df83fc00 df83fc00 f710630c f888c888 00000008 [ 1935.174007] df83fd08 00000000 f710630c f8870770 f728a0c0 f728a0c0 ffffff8d f721e180 [ 1935.174014] Call Trace: [ 1935.174021] [<f8b8cc09>] snd_usb_caiaq_midi_send+0x29/0x80 [snd_usb_caiaq] [ 1935.174034] [<f888c888>] usb_hcd_giveback_urb+0x48/0xc0 [usbcore] [ 1935.174064] [<f8870770>] ehci_urb_done+0x70/0xb0 [ehci_hcd] [ 1935.174075] [<f887153f>] qh_completions+0x24f/0x430 [ehci_hcd] [ 1935.174087] [<f8870770>] ehci_urb_done+0x70/0xb0 [ehci_hcd] [ 1935.174096] [<f887293e>] ehci_work+0x51e/0x810 [ehci_hcd] [ 1935.174103] [<c010b1e3>] sched_clock+0x13/0x40 [ 1935.174116] [<c0129199>] dequeue_task_rt+0xb9/0x1c0 [ 1935.174129] [<f8875498>] ehci_irq+0x158/0x1c0 [ehci_hcd] [ 1935.174142] [<f888cb7b>] usb_hcd_irq+0x2b/0x60 [usbcore] [ 1935.174166] [<c016a9cc>] handle_IRQ_event+0x5c/0x100 [ 1935.174178] [<c016b00a>] thread_simple_irq+0x4a/0x90 [ 1935.174185] [<c016b959>] do_irqd+0x229/0x290 [ 1935.174194] [<c016b730>] do_irqd+0x0/0x290 [ 1935.174199] [<c0141fc2>] kthread+0x42/0x70 [ 1935.174203] [<c0141f80>] kthread+0x0/0x70 [ 1935.174209] [<c0105857>] kernel_thread_helper+0x7/0x10 [ 1935.174217] ======================= [ 1935.174219] Code: 24 10 83 c4 14 c3 bf ea ff ff ff eb e3 90 83 ec 18 89 74 24 0c be ea ff ff ff 89 6c 24 14 89 d5 89 5c 24 08 89 7c 24 10 89 0c 24 <8b> 58 44 8b 7b 04 85 ff 0f 84 8e 00 00 00 8d 43 20 31 f6 89 44 [ 1935.174256] EIP: [<f8a5f6fd>] snd_rawmidi_transmit_peek+0x1d/0x100 [snd_rawmidi] SS:ESP 0068:df85be38 [ 1935.174265] ---[ end trace 9090d02b798b5496 ]--- I already asked to the Rosegarden and JACK people, but they couldn't help me: it seems to be a driver related problem, as I can happily use Rosegarden if I run JACK with the internal Intel sound card. The program versions I'm using are: jackd 0.116.1, rosegarden 1.7.2. Here's some other outputs: $ uname -a Linux din6400 2.6.24-19-rt #1 SMP PREEMPT RT Thu Aug 21 02:08:03 UTC 2008 i686 GNU/Linux $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.17. Compiled on Jan 3 2009 for kernel 2.6.24-19-rt (SMP). $ grep VERSION_STR /usr/include/alsa/version.h #define SND_LIB_VERSION_STR "1.0.17a" $ aplay -l **** Lista di PLAYBACK dispositivi hardware **** scheda 0: Intel [HDA Intel], dispositivo 0: STAC92xx Analog [STAC92xx Analog] Sottoperiferiche: 1/1 Sottoperiferica #0: subdevice #0 scheda 0: Intel [HDA Intel], dispositivo 1: STAC92xx Digital [STAC92xx Digital] Sottoperiferiche: 1/1 Sottoperiferica #0: subdevice #0 scheda 1: A1 [Audio Kontrol 1], dispositivo 0: Audio Kontrol 1 [Audio Kontrol 1] Sottoperiferiche: 1/2 Sottoperiferica #0: subdevice #0 Sottoperiferica #1: subdevice #1 $ cat /proc/interrupts CPU0 CPU1 0: 5438968 0 IO-APIC-edge timer 1: 5123 0 IO-APIC-edge i8042 8: 7 0 IO-APIC-edge rtc 9: 2 0 IO-APIC-fasteoi acpi 12: 570309 0 IO-APIC-edge i8042 14: 16584 0 IO-APIC-edge libata 15: 7712 0 IO-APIC-edge libata 16: 312032 0 IO-APIC-fasteoi i915@pci:0000:00:02.0 17: 2 0 IO-APIC-fasteoi ohci1394 18: 269358 0 IO-APIC-fasteoi uhci_hcd:usb1, ehci_hcd:usb5 19: 3999 0 IO-APIC-fasteoi uhci_hcd:usb2, HDA Intel 20: 0 0 IO-APIC-fasteoi uhci_hcd:usb3 21: 0 0 IO-APIC-fasteoi uhci_hcd:usb4 22: 7638 0 IO-APIC-fasteoi eth0 23: 0 0 IO-APIC-fasteoi sdhci:slot0 221: 31667 0 PCI-MSI-edge iwl3945 NMI: 0 0 Non-maskable interrupts LOC: 266694 2894868 Local timer interrupts RES: 1091345 1871342 Rescheduling interrupts CAL: 12287 11042 function call interrupts TLB: 1395 1581 TLB shootdowns TRM: 0 0 Thermal event interrupts SPU: 0 0 Spurious interrupts ERR: 0 MIS: 0 I do not include the jackd and rosegarden outputs becouse they are quite long, but, if you think they are useful to understand what is going on, I can send them too. Any ideas? Thank you Marco

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[alsa-devel] [PATCH] Bug in aplay/arecord --list-pcms
by Peter Stokes 12 Jan '09

12 Jan '09

Hi, I have just discovered a minor logic inversion bug in aplay/arecord --list-pcms functionality. Basically, executing "aplay --list-pcms" lists all devices capable of capture and executing "arecord --list-pcms" lists all devices capable of playback. Please find below a patch to resolve this issue. Best regards Peter Stokes ------------------------------------------------------------------------------- Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk> diff --git a/aplay/aplay.c b/aplay/aplay.c index 2e14d2a..837e46a 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -287,7 +287,7 @@ static void pcm_list(void) name = snd_device_name_get_hint(*n, "NAME"); descr = snd_device_name_get_hint(*n, "DESC"); io = snd_device_name_get_hint(*n, "IOID"); - if (io != NULL && strcmp(io, filter) == 0) + if (io != NULL && strcmp(io, filter) != 0) goto __end; printf("%s\n", name); if ((descr1 = descr) != NULL) {

2 1

[alsa-devel] [PATCH] ALSA: hda - add support for Intel DX58SO board
by Wu Fengguang 12 Jan '09

12 Jan '09

The Intel DX58SO board works fine with model ALC883_3ST_6ch_INTEL. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) --- mm.orig/sound/pci/hda/patch_realtek.c +++ mm/sound/pci/hda/patch_realtek.c @@ -8522,6 +8522,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} };

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Re: [alsa-devel] [PATCH] wss-lib: move AD1845 frequency setting into wss-lib
by Krzysztof Helt 11 Jan '09

11 Jan '09

On Sun, 11 Jan 2009 08:28:13 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > The Soundscape Elite does work, but after a reboot > I noticed that there is an oddness in that there's > a little "pop" before playing sounds with aplay and > a series of short spike pops before the sound plays > with xmms, which uses the old oss interface. > This is imperfection of early codecs. Sequences for changing sound format and rate are not yet perfect enough to avoid these "pops". Newer codecs mutes sound during such changes to avoid pops e.g. Soundscape VIVO with AD1845 does not produce "pops" before playing sounds. This is something of low priority and hard to master (guess) so do not expect solution soon. Regards, Krzysztof ---------------------------------------------------------------------- Szef przynudza? Zagraj sobie! Kliknij >> http://link.interia.pl/f200e

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[alsa-devel] The Dark Art of Sound on Linux
by A. C. Censi 11 Jan '09

11 Jan '09

This is of interest: http://glyph.twistedmatrix.com/2009/01/dark-art-of-sound-on-linux.html -- A. C. Censi accensi [em] gmail [ponto] com accensi [em] montreal [ponto] com [ponto] br

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[alsa-devel] [PATCH] wss-lib: move AD1845 frequency setting into wss-lib
by Krzysztof Helt 11 Jan '09

11 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is required to allow the sscape driver to autodetect installed codec. Also, do not create a timer if detected codec has no hardware timer (e.g. AD1848). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> Cc: Rene Herman --- Ted, please test if the patch helps to make your Soundscape Elite card working. If not the patch is required anyway as your last feedback showed that frequency setting is incorrect in the sscape driver if the codec is not the AD1845. sound/isa/sscape.c | 113 +++------------------------------------------- sound/isa/wss/wss_lib.c | 40 +++++++++++++++++ 2 files changed, 48 insertions(+), 105 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 48a16d8..bc44916 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -129,9 +129,6 @@ enum GA_REG { #define DMA_8BIT 0x80 -#define AD1845_FREQ_SEL_MSB 0x16 -#define AD1845_FREQ_SEL_LSB 0x17 - enum card_type { SSCAPE, SSCAPE_PNP, @@ -955,82 +952,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l /* - * Override for the CS4231 playback format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_playback_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the PLAYBACK_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* - * Override for the CS4231 capture format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_capture_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the CAPTURE_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* * Create an AD1845 PCM subdevice on the SoundScape. The AD1845 * is very much like a CS4231, with a few extra bits. We will * try to support at least some of the extra bits by overriding @@ -1055,11 +976,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -#define AD1845_FREQ_SEL_ENABLE 0x08 - -#define AD1845_PWR_DOWN_CTRL 0x1b -#define AD1845_CRYS_CLOCK_SEL 0x1d - /* * It turns out that the PLAYBACK_ENABLE bit is set * by the lowlevel driver ... @@ -1074,7 +990,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ if (sscape->type != SSCAPE_VIVO) { - int val; /* * The input clock frequency on the SoundScape must * be 14.31818 MHz, because we must set this register @@ -1082,22 +997,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + snd_wss_out(chip, AD1845_CLOCK, 0x20); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); - /* - * More custom configuration: - * a) select "mode 2" and provide a current drive of 8mA - * b) enable frequency selection (for capture/playback) - */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, - CS4231_MODE2 | 0x10); - val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL); - snd_wss_out(chip, AD1845_PWR_DOWN_CTRL, - val | AD1845_FREQ_SEL_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); } err = snd_wss_pcm(chip, 0, &pcm); @@ -1113,11 +1016,13 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "for AD1845 chip\n"); goto _error; } - err = snd_wss_timer(chip, 0, NULL); - if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " - "for AD1845 chip\n"); - goto _error; + if (chip->hardware != WSS_HW_AD1848) { + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No timer device " + "for AD1845 chip\n"); + goto _error; + } } if (sscape->type != SSCAPE_VIVO) { @@ -1128,8 +1033,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "MIDI mixer control\n"); goto _error; } - chip->set_playback_format = ad1845_playback_format; - chip->set_capture_format = ad1845_capture_format; } strcpy(card->driver, "SoundScape"); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2..13299ae 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -646,6 +646,24 @@ static void snd_wss_playback_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the playback stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (pdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -690,6 +708,24 @@ static void snd_wss_capture_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the capture stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_REC_FORMAT, (cdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -1314,6 +1350,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } + /* enable fine grained frequency selection */ + if (chip->hardware == WSS_HW_AD1845) + chip->image[AD1845_PWR_DOWN] = 8; + ptr = (unsigned char *) &chip->image; regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); -- 1.5.2.2 ---------------------------------------------------------------------- Promocja w Speak Up. 3 miesiace angielskiego gratis. Sprawdz teraz i wypelnij formularz! >> http://link.interia.pl/f2019

2 2

[alsa-devel] [patch] Indigo DJx
by carmen 11 Jan '09

11 Jan '09

the old Indigo DJ driver worked after a few tweaks... tested w/ jack, cmus, mplayer mapping/virtual-outs seem good.. this is a diff on alsa-kernel.git. copied the old firmware and amazed it works, although something seems wonky with the samplerate registers. going to email echo about an updated firmware .. no noise bursts or ear destruction tho!

3 4

Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 11 Jan '09

11 Jan '09

At Sat, 10 Jan 2009 09:34:57 -0500, Kyler Laird wrote: > > On Fri, Jan 09, 2009 at 05:14:45PM +0100, Takashi Iwai wrote: > > > Could you try test/cf74-fix branch on sound git tree? > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/cf74-fix > > That didn't work for me. I see: > HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ > 22 > HDA Intel 0000:00:1b.0: setting latency timer to 64 > input: HDA Intel at 0xe8640000 irq 22 Line In at Ext Rear Jack > as /class/input/input7 > input: HDA Intel at 0xe8640000 irq 22 Mic at Ext Front Jack as > /class/input/input8 > input: HDA Intel at 0xe8640000 irq 22 Mic at Ext Rear Jack as > /class/input/input9 > input: HDA Intel at 0xe8640000 irq 22 Speaker at Ext Rear Jack > as /class/input/input10 > input: HDA Intel at 0xe8640000 irq 22 HP Out at Ext Front Jack > as /class/input/input11 These are all OK. The new messages regarding input device are for jack-sensing stuff. Could you run alsa-info.sh on the latest kernel, too? Also, to be sure, try to put a printk in the quirk for CF-74 (disabling hp_detect) to confirm it's activated. thanks, Takashi

1 0

[alsa-devel] [PATCH 0/2] ASoC: MioA701 board
by Robert Jarzmik 09 Jan '09

09 Jan '09

Hi, These 2 patches have been already reviewed in April 2008 and have been siting in Mark's asoc-v2 tree since then. They have been ported back to asoc-v1 for mainline submission. Please consider them for review if need be and scheduling for the next queue. The serie was build upon Takashi's topic/asoc branch, commit "ASoC: Clean up kerneldoc warnings" of id ac11a2b35cc25c77d28218aaf60e7f7c6c7ee5d3. Cheers. -- Robert Robert Jarzmik (2): Add initial support of Mitac mioa701 device SoC. Add initial support of Mitac mioa701 master volume. sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_masterctrl.c | 243 +++++++++++++++ sound/soc/pxa/mioa701_wm9713.c | 578 ++++++++++++++++++++++++++++++++++++ 4 files changed, 832 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/mioa701_masterctrl.c create mode 100644 sound/soc/pxa/mioa701_wm9713.c

4 11

Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 09 Jan '09

09 Jan '09

At Fri, 9 Jan 2009 10:21:21 -0500, Kyler Laird wrote: > > On Fri, Jan 09, 2009 at 08:51:58AM +0100, Takashi Iwai wrote: > > > Weird. I thought this path is active only when hp_detect != 0. > > Could you check whether it's really correctly patched, e.g. by the > > patch like below? > > That patch works. Thanks! > > > It seems applying my previous patch on 2.6.28 puts the code into a > > wrong place. > > I haven't succeeded in getting either of your recent patches to work > against 2.6.28-git13. I'll work on it more later. The previous patch was for 2.6.28 while the first one was for git. Maybe just because of a line difference. Could you try test/cf74-fix branch on sound git tree? git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/cf74-fix Takashi

1 0

[alsa-devel] alsa-plugin: rate-libsamplerate, buffer shrinkage problem
by John L. Utz III 09 Jan '09

09 Jan '09

Hi; Our legacy product code is tormenting me again. :-( i have a 3rd party sound-fiddling library that only works with an 8k buffer. the company that wrote it is long gone. i cant change this. i have a enourmous array of content that is coded at 44.1K. i cant change this. i have to use a cheap soundblaster audigy card (ca0106) that only supports 48K. it sounds really good at 48K, so it seems like a winner. the samplerate plugin is almost perfect for helping me successfully use the 48K only card with our 44.1K content. *almost*. :-( if i use the samplerate plugin thusly in my CA0106.conf: CA0106.pcm.front.0 { @args [ CARD ] @args.CARD { type string } # type hw # card $CARD type rate slave { pcm "hw:0,0" rate 48000 } converter "samplerate" } I no longer get a 8192 byte buffer, i get a 7526 byte buffer. :-( snd_pcm_hw_params_set_buffer_size_near() fails if i try and set it to 8192, but it accepts 7526 or 15092. I *assume* that either the samplerate plugin or libsamplerate are consuming part of the buffer for the resample job. Not at all unreasonable. But it seems like i should be able to bump up the internal working buffer size so that i get my 8192 byte submittal buffer back. Does anybody have any suggestions? Is there a flag or config entry that i have missed? Or do i need to go hack something? That would be fine, but it would be nice to know where to look. :-) Here is a succinct example of the problem, first i call speaker-test without a rate argument and then with the rate argument: test-gntopxsg ~ # speaker-test -c2 -Dfront -t sin speaker-test 1.0.17 Playback device is front Stream parameters are 48000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 48000Hz (requested 48000Hz) Buffer size range from 32 to 16384 Period size range from 15 to 8193 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 test-gntopxsg ~ # speaker-test -c2 -Dfront -t sin -r44100 speaker-test 1.0.17 Playback device is front Stream parameters are 44100Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 44100Hz (requested 44100Hz) Buffer size range from 29 to 15052 Period size range from 14 to 7527 Using max buffer size 15052 Periods = 4 was set period_size = 3763 was set buffer_size = 15052 tnx in advance for any help you all can provide! johnu

2 1

[alsa-devel] [PATCH 0/7] ASoc updates for -next
by Mark Brown 09 Jan '09

09 Jan '09

The following changes since commit 227b4dc6432d271eecd0ff0aefe6f0897ec47397: Mark Brown (1): ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Ian Molton (1): ASoC: dapm: Allow explictly named mixer controls Mark Brown (4): Merge branch 'asoc' of git://git.mnementh.co.uk/linux-2.6-im into for-tiwai ASoC: Constify pin names for DAPM pin status APIs ASoC: Add jack reporting interface ASoC: Implement WM8350 headphone jack detection Peter Ujfalusi (3): ASoC: Merge the soc_value_enum to soc_enum struct ASoC: TWL4030: Change the soc_value_enum back to soc_enum ASoC: TWL4030: Module unloading fix Documentation/sound/alsa/soc/dapm.txt | 3 + include/linux/mfd/wm8350/audio.h | 1 + include/sound/soc-dapm.h | 21 ++++- include/sound/soc.h | 47 ++++++++---- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/codecs/twl4030.c | 12 ++- sound/soc/codecs/wm8350.c | 116 +++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 8 ++ sound/soc/soc-core.c | 37 +-------- sound/soc/soc-dapm.c | 137 +++++++++++---------------------- sound/soc/soc-jack.c | 138 +++++++++++++++++++++++++++++++++ 12 files changed, 372 insertions(+), 151 deletions(-) create mode 100644 sound/soc/soc-jack.c

3 12

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 09 Jan '09

09 Jan '09

The following changes since commit 227b4dc6432d271eecd0ff0aefe6f0897ec47397: Mark Brown (1): ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ian Molton (2): ASoC: dapm: Allow explictly named mixer controls ASoC: cleanup duplicated code. Mark Brown (4): ASoC: Constify pin names for DAPM pin status APIs ASoC: Add jack reporting interface ASoC: Implement WM8350 headphone jack detection Merge branch 'for-2.6.29' into for-2.6.30 Peter Ujfalusi (3): ASoC: Merge the soc_value_enum to soc_enum struct ASoC: TWL4030: Change the soc_value_enum back to soc_enum ASoC: TWL4030: Module unloading fix Documentation/sound/alsa/soc/dapm.txt | 3 + include/linux/mfd/wm8350/audio.h | 1 + include/sound/soc-dapm.h | 21 ++++- include/sound/soc.h | 49 ++++++++---- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/codecs/ad1980.c | 17 +---- sound/soc/codecs/ak4535.c | 18 +---- sound/soc/codecs/ssm2602.c | 18 +---- sound/soc/codecs/tlv320aic23.c | 21 +----- sound/soc/codecs/tlv320aic3x.c | 19 +---- sound/soc/codecs/twl4030.c | 31 ++----- sound/soc/codecs/uda134x.c | 50 ++++-------- sound/soc/codecs/uda1380.c | 18 +---- sound/soc/codecs/wm8350.c | 134 ++++++++++++++++++++++++++++---- sound/soc/codecs/wm8350.h | 8 ++ sound/soc/codecs/wm8510.c | 19 +---- sound/soc/codecs/wm8580.c | 17 +---- sound/soc/codecs/wm8728.c | 18 +---- sound/soc/codecs/wm8731.c | 19 +---- sound/soc/codecs/wm8750.c | 18 +---- sound/soc/codecs/wm8753.c | 18 +---- sound/soc/codecs/wm8900.c | 19 +---- sound/soc/codecs/wm8903.c | 18 +---- sound/soc/codecs/wm8971.c | 18 +---- sound/soc/codecs/wm8990.c | 18 +---- sound/soc/codecs/wm9712.c | 18 +---- sound/soc/codecs/wm9713.c | 18 +---- sound/soc/soc-core.c | 68 ++++++++-------- sound/soc/soc-dapm.c | 137 +++++++++++---------------------- sound/soc/soc-jack.c | 138 +++++++++++++++++++++++++++++++++ 31 files changed, 461 insertions(+), 511 deletions(-) create mode 100644 sound/soc/soc-jack.c

2 1

[alsa-devel] ASoC updates for 2.6.29
by Mark Brown 09 Jan '09

09 Jan '09

The following changes since commit 227b4dc6432d271eecd0ff0aefe6f0897ec47397: Mark Brown (1): ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Peter Ujfalusi (3): ASoC: Merge the soc_value_enum to soc_enum struct ASoC: TWL4030: Change the soc_value_enum back to soc_enum ASoC: TWL4030: Module unloading fix include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 15 +-------- sound/soc/codecs/twl4030.c | 12 ++++--- sound/soc/soc-core.c | 37 +------------------- sound/soc/soc-dapm.c | 80 +++----------------------------------------- 5 files changed, 16 insertions(+), 130 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - create hda_codec.control_mutex for kcontrol->private_value
by Wu Fengguang 09 Jan '09

09 Jan '09

On Fri, Jan 09, 2009 at 08:18:47AM +0100, Takashi Iwai wrote: > At Fri, 9 Jan 2009 13:06:21 +0800, > Wu Fengguang wrote: > > > > On Thu, Jan 08, 2009 at 04:20:37PM +0100, Ingo Molnar wrote: > > > > > > FYI, -tip testing found this new lockdep workqueue locking assert > > > triggered by the Intel HDA sound driver: > > > > > > [ 30.637612] alloc kstat_irqs on cpu 0 node 0 > > > [ 30.647029] IOAPIC[0]: Set routing entry (2-22 -> 0x71 -> IRQ 22 Mode:1 Active:1) > > > [ 30.654533] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > > > [ 30.661782] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > > [ 30.667801] chipset global capabilities = 0x4401 > > > [ 30.701090] codec_mask = 0x4 > > > [ 30.701090] hda_intel: codec #2 probed OK > > > [ 30.709090] hda-codec: No codec parser is available > > > [ 30.714396] hda-intel: no codecs initialized > > > [ 30.718689] > > > [ 30.718689] ============================================= > > > [ 30.718689] [ INFO: possible recursive locking detected ] > > > [ 30.718689] 2.6.28-tip #2 > > > [ 30.718689] --------------------------------------------- > > > [ 30.718689] events/0/7 is trying to acquire lock: > > > [ 30.718689] (events){--..}, at: [<ffffffff80281050>] flush_workqueue+0x0/0x90 > > > [ 30.718689] > > > [ 30.718689] but task is already holding lock: > > > [ 30.718689] (events){--..}, at: [<ffffffff8028076a>] run_workqueue+0x14a/0x240 > > > [ 30.718689] > > > [ 30.718689] other info that might help us debug this: > > > [ 30.718689] 2 locks held by events/0/7: > > > [ 30.718689] #0: (events){--..}, at: [<ffffffff8028076a>] run_workqueue+0x14a/0x240 > > > [ 30.718689] #1: (&wfc.work){--..}, at: [<ffffffff8028076a>] run_workqueue+0x14a/0x240 > > > > I didn't see this bug, but ran into another one with linux-next 20090102: > > > > [ 844.111765] ALSA sound/pci/hda/hda_codec.c:882: hda_codec_cleanup_stream: NID=0x2 > > [ 844.111915] > > [ 844.111916] ======================================================= > > [ 844.112018] [ INFO: possible circular locking dependency detected ] > > [ 844.112076] 2.6.28-next-20090102 #33 > > [ 844.112124] ------------------------------------------------------- > > [ 844.112181] mplayer/3151 is trying to acquire lock: > > [ 844.112235] (&pcm->open_mutex){--..}, at: [<ffffffffa004ced3>] snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112361] > > [ 844.112362] but task is already holding lock: > > [ 844.112458] (&mm->mmap_sem){----}, at: [<ffffffff810c0252>] sys_munmap+0x42/0x80 > > [ 844.112576] > > [ 844.112577] which lock already depends on the new lock. > > [ 844.112578] > > [ 844.112712] > > [ 844.112713] the existing dependency chain (in reverse order) is: > > [ 844.112781] > > [ 844.112781] -> #3 (&mm->mmap_sem){----}: > > [ 844.112781] [<ffffffff8107097b>] __lock_acquire+0x122b/0x1ae0 > > [ 844.112781] [<ffffffff810712c9>] lock_acquire+0x99/0xd0 > > [ 844.112781] [<ffffffff810b7a43>] might_fault+0x73/0xa0 > > [ 844.112781] [<ffffffffa006f44d>] snd_hda_mixer_amp_tlv+0x6d/0xf0 [snd_hda_codec] > > [ 844.112781] [<ffffffffa0093f80>] alc_cap_vol_tlv+0x70/0xa0 [snd_hda_codec_realtek] > > [ 844.112781] [<ffffffffa0012300>] snd_ctl_tlv_ioctl+0x180/0x1b0 [snd] > > [ 844.112781] [<ffffffffa0013a27>] snd_ctl_ioctl+0x927/0xf80 [snd] > > [ 844.112781] [<ffffffff810ea801>] vfs_ioctl+0x31/0xa0 > > [ 844.112781] [<ffffffff810ea8e6>] do_vfs_ioctl+0x76/0x4a0 > > [ 844.112781] [<ffffffff810ead5a>] sys_ioctl+0x4a/0x80 > > [ 844.112781] [<ffffffff8100c67a>] system_call_fastpath+0x16/0x1b > > [ 844.112781] [<ffffffffffffffff>] 0xffffffffffffffff > > [ 844.112781] > > [ 844.112781] -> #2 (&codec->spdif_mutex){--..}: > > [ 844.112781] [<ffffffff8107097b>] __lock_acquire+0x122b/0x1ae0 > > [ 844.112781] [<ffffffff810712c9>] lock_acquire+0x99/0xd0 > > [ 844.112781] [<ffffffff814344e8>] mutex_lock_nested+0xe8/0x370 > > [ 844.112781] [<ffffffffa006e54b>] snd_hda_multi_out_dig_open+0x2b/0x60 [snd_hda_codec] > > [ 844.112781] [<ffffffffa00d2649>] intel_hdmi_playback_pcm_open+0x49/0x60 [snd_hda_codec_intelhdmi] > > [ 844.112781] [<ffffffffa007fff9>] azx_pcm_open+0x209/0x280 [snd_hda_intel] > > [ 844.112781] [<ffffffffa004d838>] snd_pcm_open_substream+0x58/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffffa004d9e7>] snd_pcm_open+0x137/0x260 [snd_pcm] > > [ 844.112781] [<ffffffffa004db7c>] snd_pcm_playback_open+0x2c/0x40 [snd_pcm] > > [ 844.112781] [<ffffffffa000e86e>] snd_open+0x9e/0x180 [snd] > > [ 844.112781] [<ffffffff810df407>] chrdev_open+0xe7/0x1d0 > > [ 844.112781] [<ffffffff810da06e>] __dentry_open+0xce/0x360 > > [ 844.112781] [<ffffffff810da407>] nameidata_to_filp+0x57/0x70 > > [ 844.112781] [<ffffffff810e909e>] do_filp_open+0x20e/0x970 > > [ 844.112781] [<ffffffff810d9ec8>] do_sys_open+0x78/0x100 > > [ 844.112781] [<ffffffff810d9f7b>] sys_open+0x1b/0x20 > > [ 844.112781] [<ffffffff8100c67a>] system_call_fastpath+0x16/0x1b > > [ 844.112781] [<ffffffffffffffff>] 0xffffffffffffffff > > [ 844.112781] > > [ 844.112781] -> #1 (&chip->open_mutex){--..}: > > [ 844.112781] [<ffffffff8107097b>] __lock_acquire+0x122b/0x1ae0 > > [ 844.112781] [<ffffffff810712c9>] lock_acquire+0x99/0xd0 > > [ 844.112781] [<ffffffff814344e8>] mutex_lock_nested+0xe8/0x370 > > [ 844.112781] [<ffffffffa007fe38>] azx_pcm_open+0x48/0x280 [snd_hda_intel] > > [ 844.112781] [<ffffffffa004d838>] snd_pcm_open_substream+0x58/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffffa004d9e7>] snd_pcm_open+0x137/0x260 [snd_pcm] > > [ 844.112781] [<ffffffffa004db7c>] snd_pcm_playback_open+0x2c/0x40 [snd_pcm] > > [ 844.112781] [<ffffffffa000e86e>] snd_open+0x9e/0x180 [snd] > > [ 844.112781] [<ffffffff810df407>] chrdev_open+0xe7/0x1d0 > > [ 844.112781] [<ffffffff810da06e>] __dentry_open+0xce/0x360 > > [ 844.112781] [<ffffffff810da407>] nameidata_to_filp+0x57/0x70 > > [ 844.112781] [<ffffffff810e909e>] do_filp_open+0x20e/0x970 > > [ 844.112781] [<ffffffff810d9ec8>] do_sys_open+0x78/0x100 > > [ 844.112781] [<ffffffff810d9f7b>] sys_open+0x1b/0x20 > > [ 844.112781] [<ffffffff8100c67a>] system_call_fastpath+0x16/0x1b > > [ 844.112781] [<ffffffffffffffff>] 0xffffffffffffffff > > [ 844.112781] > > [ 844.112781] -> #0 (&pcm->open_mutex){--..}: > > [ 844.112781] [<ffffffff81070b1e>] __lock_acquire+0x13ce/0x1ae0 > > [ 844.112781] [<ffffffff810712c9>] lock_acquire+0x99/0xd0 > > [ 844.112781] [<ffffffff814344e8>] mutex_lock_nested+0xe8/0x370 > > [ 844.112781] [<ffffffffa004ced3>] snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffff810dd505>] __fput+0xd5/0x230 > > [ 844.112781] [<ffffffff810dd67d>] fput+0x1d/0x30 > > [ 844.112781] [<ffffffff810bed4f>] remove_vma+0x4f/0x90 > > [ 844.112781] [<ffffffff810c01c2>] do_munmap+0x342/0x390 > > [ 844.112781] [<ffffffff810c0260>] sys_munmap+0x50/0x80 > > [ 844.112781] [<ffffffff8100c67a>] system_call_fastpath+0x16/0x1b > > [ 844.112781] [<ffffffffffffffff>] 0xffffffffffffffff > > [ 844.112781] > > [ 844.112781] other info that might help us debug this: > > [ 844.112781] > > [ 844.112781] 1 lock held by mplayer/3151: > > [ 844.112781] #0: (&mm->mmap_sem){----}, at: [<ffffffff810c0252>] sys_munmap+0x42/0x80 > > [ 844.112781] > > [ 844.112781] stack backtrace: > > [ 844.112781] Pid: 3151, comm: mplayer Not tainted 2.6.28-next-20090102 #33 > > [ 844.112781] Call Trace: > > [ 844.112781] [<ffffffff8106f2a0>] print_circular_bug_tail+0xe0/0xf0 > > [ 844.112781] [<ffffffff81070b1e>] __lock_acquire+0x13ce/0x1ae0 > > [ 844.112781] [<ffffffff8106e916>] ? mark_held_locks+0x56/0xa0 > > [ 844.112781] [<ffffffff810abcf9>] ? free_hot_cold_page+0x1b9/0x2e0 > > [ 844.112781] [<ffffffff8106ec1d>] ? trace_hardirqs_on+0xd/0x10 > > [ 844.112781] [<ffffffff810712c9>] lock_acquire+0x99/0xd0 > > [ 844.112781] [<ffffffffa004ced3>] ? snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffff814344e8>] mutex_lock_nested+0xe8/0x370 > > [ 844.112781] [<ffffffffa004ced3>] ? snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffffa004ced3>] ? snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffffa004ced3>] snd_pcm_release+0x43/0xd0 [snd_pcm] > > [ 844.112781] [<ffffffff810dd505>] __fput+0xd5/0x230 > > [ 844.112781] [<ffffffff810dd67d>] fput+0x1d/0x30 > > [ 844.112781] [<ffffffff810bed4f>] remove_vma+0x4f/0x90 > > [ 844.112781] [<ffffffff810c01c2>] do_munmap+0x342/0x390 > > [ 844.112781] [<ffffffff810c0260>] sys_munmap+0x50/0x80 > > [ 844.112781] [<ffffffff8100c67a>] system_call_fastpath+0x16/0x1b > > [ 844.119376] ALSA sound/pci/hda/hda_codec.c:882: hda_codec_cleanup_stream: NID=0x2 > > > > Which can be fixed by the following patch. > > This alone isn't enough. You need to replace all codec->spdif_mutex > in the control get/put/info/tlv callbacks in hda_codec.c. Otherwise > there is no meaning of the mutex. The mutex is to protect the rewrite > of kcontrol->private_value dynamically. > > Could you fix and repost? Ah sure. Here is the updated patch. Thanks, Fengguang --- ALSA: hda - create hda_codec.control_mutex for kcontrol->private_value Fix the following lockdep warning by not reusing the hda_codec.spdif_mutex. ALSA sound/pci/hda/hda_codec.c:882: hda_codec_cleanup_stream: NID=0x2 ======================================================= [ INFO: possible circular locking dependency detected ] 2.6.28-next-20090102 #33 ------------------------------------------------------- mplayer/3151 is trying to acquire lock: (&pcm->open_mutex){--..}, at: [<ffffffffa004ced3>] snd_pcm_release+0x43/0xd0 [snd_pcm] but task is already holding lock: (&mm->mmap_sem){----}, at: [<ffffffff810c0252>] sys_munmap+0x42/0x80 which lock already depends on the new lock. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_codec.c | 25 +++++++++++++------------ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_realtek.c | 12 ++++++------ 3 files changed, 20 insertions(+), 18 deletions(-) --- mm.orig/sound/pci/hda/hda_codec.h +++ mm/sound/pci/hda/hda_codec.h @@ -771,6 +771,7 @@ struct hda_codec { struct hda_cache_rec cmd_cache; /* cache for other commands */ struct mutex spdif_mutex; + struct mutex control_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ --- mm.orig/sound/pci/hda/hda_codec.c +++ mm/sound/pci/hda/hda_codec.c @@ -735,6 +735,7 @@ int /*__devinit*/ snd_hda_codec_new(stru codec->bus = bus; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); @@ -1418,12 +1419,12 @@ int snd_hda_mixer_bind_switch_get(struct unsigned long pval; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); @@ -1435,7 +1436,7 @@ int snd_hda_mixer_bind_switch_put(struct unsigned long pval; int i, indices, err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT; for (i = 0; i < indices; i++) { @@ -1447,7 +1448,7 @@ int snd_hda_mixer_bind_switch_put(struct change |= err; } kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); @@ -1462,12 +1463,12 @@ int snd_hda_mixer_bind_ctls_info(struct struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->info(kcontrol, uinfo); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); @@ -1479,12 +1480,12 @@ int snd_hda_mixer_bind_ctls_get(struct s struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->get(kcontrol, ucontrol); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); @@ -1497,7 +1498,7 @@ int snd_hda_mixer_bind_ctls_put(struct s unsigned long *vals; int err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; for (vals = c->values; *vals; vals++) { kcontrol->private_value = *vals; @@ -1507,7 +1508,7 @@ int snd_hda_mixer_bind_ctls_put(struct s change |= err; } kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); @@ -1519,12 +1520,12 @@ int snd_hda_mixer_bind_tlv(struct snd_kc struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->tlv(kcontrol, op_flag, size, tlv); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv); --- mm.orig/sound/pci/hda/patch_realtek.c +++ mm/sound/pci/hda/patch_realtek.c @@ -1502,11 +1502,11 @@ static int alc_cap_vol_info(struct snd_k struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1517,11 +1517,11 @@ static int alc_cap_vol_tlv(struct snd_kc struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1537,11 +1537,11 @@ static int alc_cap_getput_caller(struct unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], 3, 0, HDA_INPUT); err = func(kcontrol, ucontrol); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; }

2 1

Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 09 Jan '09

09 Jan '09

At Thu, 8 Jan 2009 13:22:41 -0500, Kyler Laird wrote: > > On Thu, Jan 08, 2009 at 06:54:03PM +0100, Takashi Iwai wrote: > > > Could you add dump_stack() in stac92xx_unsol_event() and check the > > code path? > > Jan 8 13:19:12 ruff kernel: HDA Intel 0000:00:1b.0: PCI INT A disabled > Jan 8 13:19:12 ruff kernel: HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > Jan 8 13:19:12 ruff kernel: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > Jan 8 13:19:12 ruff kernel: HDA Intel 0000:00:1b.0: setting latency timer to 64 > Jan 8 13:19:12 ruff kernel: (Kyler) stac92xx_unsol_event: 3221225472 > Jan 8 13:19:12 ruff kernel: Pid: 13602, comm: modprobe Not tainted 2.6.28 #1 > Jan 8 13:19:12 ruff kernel: Call Trace: > Jan 8 13:19:12 ruff kernel: [<e0a30af3>] stac92xx_unsol_event+0x33/0x380 [snd_hda_intel] > Jan 8 13:19:12 ruff kernel: [<e0a303b0>] stac92xx_init+0x120/0x490 [snd_hda_intel] Weird. I thought this path is active only when hp_detect != 0. Could you check whether it's really correctly patched, e.g. by the patch like below? It seems applying my previous patch on 2.6.28 puts the code into a wrong place. Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b77f330..9982e4b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3745,6 +3745,7 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { + printk(KERN_DEBUG "XXX stac92xx_init -- hp_detect is ON!\n"); /* Enable unsolicited responses on the HP widget */ for (i = 0; i < cfg->hp_outs; i++) enable_pin_detect(codec, cfg->hp_pins[i], @@ -4129,6 +4130,11 @@ static int patch_stac9200(struct hda_codec *codec) return err; } + if (spec->board_config == STAC_9200_PANASONIC) { + printk(KERN_DEBUG "XXX panasonic quirk enabled\n"); + spec->hp_detect = 0; + } + codec->patch_ops = stac92xx_patch_ops; return 0;

1 0

[alsa-devel] PATCH: Cleanup duplicate code
by Ian Molton 09 Jan '09

09 Jan '09

Hi folks, I noticed this function has been copied, 'cookie cutter' style in a lot of codec drivers. Here is a patch that cleans it up. Also available in my git repo: git://git.mnementh.co.uk/linux-2.6-im.git asoc From 8cf35a0f438c70eb7dc3334745ce4658b69f9e8a Mon Sep 17 00:00:00 2001 From: Ian Molton <ian(a)mnementh.co.uk> Date: Fri, 9 Jan 2009 00:23:21 +0000 Subject: [PATCH] ASoC: cleanup duplicated code. Many codec drivers were implementing cookie-cutter copies of the function that adds kcontrols to the codec. This patch moves this code to a common function snd_soc_add_controls() in soc-core.c and updates all drivers using copies of this function to use the new common version. Signed-off-by: Ian Molton <ian(a)mnementh.co.uk> --- include/sound/soc.h | 2 + sound/soc/codecs/ad1980.c | 17 +------------ sound/soc/codecs/ak4535.c | 18 +------------ sound/soc/codecs/ssm2602.c | 18 +------------ sound/soc/codecs/tlv320aic23.c | 21 +--------------- sound/soc/codecs/tlv320aic3x.c | 19 +------------- sound/soc/codecs/twl4030.c | 19 +------------- sound/soc/codecs/uda134x.c | 50 ++++++++++++--------------------------- sound/soc/codecs/uda1380.c | 18 +------------ sound/soc/codecs/wm8350.c | 18 +------------ sound/soc/codecs/wm8510.c | 19 +------------- sound/soc/codecs/wm8580.c | 17 +------------ sound/soc/codecs/wm8728.c | 18 +------------ sound/soc/codecs/wm8731.c | 19 +------------- sound/soc/codecs/wm8750.c | 18 +------------ sound/soc/codecs/wm8753.c | 18 +------------ sound/soc/codecs/wm8900.c | 19 +------------- sound/soc/codecs/wm8903.c | 18 +------------ sound/soc/codecs/wm8971.c | 18 +------------ sound/soc/codecs/wm8990.c | 18 +------------ sound/soc/codecs/wm9712.c | 18 +------------ sound/soc/codecs/wm9713.c | 18 +------------ sound/soc/soc-core.c | 23 ++++++++++++++++++ 23 files changed, 81 insertions(+), 360 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9b930d3..7223101 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -203,6 +203,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name); +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 73fdbb4..c3c5d0e 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; -/* add non dapm controls */ -static int ad1980_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, snd_soc_cnew( - &ad1980_snd_ac97_controls[i], codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - ad1980_add_controls(codec); + snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + ARRAY_SIZE(ad1980_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8..f17c363 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = { SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), }; -/* add non dapm controls */ -static int ak4535_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Mono 1 Mixer */ static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), @@ -510,7 +495,8 @@ static int ak4535_init(struct snd_soc_device *socdev) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ak4535_add_controls(codec); + snd_soc_add_controls(codec, ak4535_snd_controls, + ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac3736..ec7fe3b 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]), SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; -/* add non dapm controls */ -static int ssm2602_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), @@ -622,7 +607,8 @@ static int ssm2602_init(struct snd_soc_device *socdev) APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); - ssm2602_add_controls(codec); + snd_soc_add_controls(codec, ssm2602_snd_controls, + ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea00..a0e47c1 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -/* add non dapm controls */ -static int tlv320aic23_add_controls(struct snd_soc_codec *codec) -{ - - int err, i; - - for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tlv320aic23_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; - -} - /* PGA Mixer controls for Line and Mic switch */ static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), @@ -718,7 +700,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); - tlv320aic23_add_controls(codec); + snd_soc_add_controls(codec, tlv320aic23_snd_controls, + ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..36ab019 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -311,22 +311,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; -/* add non dapm controls */ -static int aic3x_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic3x_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); @@ -1224,7 +1208,8 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); - aic3x_add_controls(codec); + snd_soc_add_controls(codec, aic3x_snd_controls, + ARRAY_SIZE(aic3x_snd_controls)); aic3x_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd0f338..253063f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -670,22 +670,6 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 0, 3, 5, 0, input_gain_tlv), }; -/* add non dapm controls */ -static int twl4030_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&twl4030_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Left channel inputs */ SND_SOC_DAPM_INPUT("MAINMIC"), @@ -1233,7 +1217,8 @@ static int twl4030_init(struct snd_soc_device *socdev) /* power on device */ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - twl4030_add_controls(codec); + snd_soc_add_controls(codec, twl4030_snd_controls, + ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064..277825d 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,39 +431,6 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; -static int uda134x_add_controls(struct snd_soc_codec *codec) -{ - int err, i, n; - const struct snd_kcontrol_new *ctrls; - struct uda134x_platform_data *pd = codec->control_data; - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - n = ARRAY_SIZE(uda1340_snd_controls); - ctrls = uda1340_snd_controls; - break; - case UDA134X_UDA1341: - n = ARRAY_SIZE(uda1341_snd_controls); - ctrls = uda1341_snd_controls; - break; - default: - printk(KERN_ERR "%s unkown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - for (i = 0; i < n; i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ctrls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -572,7 +539,22 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = uda134x_add_controls(codec); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ARRAY_SIZE(uda1340_snd_controls)); + break; + case UDA134X_UDA1341: + ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ARRAY_SIZE(uda1341_snd_controls)); + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + if (ret < 0) { printk(KERN_ERR "UDA134X: failed to register controls\n"); goto pcm_err; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf084..a957b43 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -271,21 +271,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; -/* add non dapm controls */ -static int uda1380_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); @@ -675,7 +660,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) } /* uda1380 init */ - uda1380_add_controls(codec); + snd_soc_add_controls(codec, uda1380_snd_controls, + ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d4..db00b6f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -775,21 +775,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Beep", NULL, "IN3R PGA"}, }; -static int wm8350_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8350_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static int wm8350_add_widgets(struct snd_soc_codec *codec) { int ret; @@ -1387,7 +1372,8 @@ static int wm8350_probe(struct platform_device *pdev) return ret; } - wm8350_add_controls(codec); + snd_soc_add_controls(codec, wm8350_snd_controls, + ARRAY_SIZE(wm8350_snd_controls)); wm8350_add_widgets(codec); wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238..abe7cce 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; -/* add non dapm controls */ -static int wm8510_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8510_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Speaker Output Mixer */ static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), @@ -656,7 +640,8 @@ static int wm8510_init(struct snd_soc_device *socdev) /* power on device */ codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8510_add_controls(codec); + snd_soc_add_controls(codec, wm8510_snd_controls, + ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e58..9b75a37 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -330,20 +330,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), }; -/* Add non-DAPM controls */ -static int wm8580_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8580_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), @@ -866,7 +852,8 @@ static int wm8580_init(struct snd_soc_device *socdev) goto pcm_err; } - wm8580_add_controls(codec); + snd_soc_add_controls(codec, wm8580_snd_controls, + ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b1198..defa310 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), }; -static int wm8728_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8728_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* * DAPM controls. */ @@ -330,7 +315,8 @@ static int wm8728_init(struct snd_soc_device *socdev) /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8728_add_controls(codec); + snd_soc_add_controls(codec, wm8728_snd_controls, + ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f..96d6e1a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -129,22 +129,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), }; -/* add non dapm controls */ -static int wm8731_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), @@ -543,7 +527,8 @@ static int wm8731_init(struct snd_soc_device *socdev) reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); - wm8731_add_controls(codec); + snd_soc_add_controls(codec, wm8731_snd_controls, + ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa6..1578569 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), }; -/* add non dapm controls */ -static int wm8750_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -816,7 +801,8 @@ static int wm8750_init(struct snd_soc_device *socdev) reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); - wm8750_add_controls(codec); + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50..7283178 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]), SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; -/* add non dapm controls */ -static int wm8753_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1603,7 +1588,8 @@ static int wm8753_init(struct snd_soc_device *socdev) reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); - wm8753_add_controls(codec); + snd_soc_add_controls(codec, wm8753_snd_controls, + ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de1..1e08d4f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1, }; -/* add non dapm controls */ -static int wm8900_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8900_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new wm8900_dapm_loutput2_control = SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0); @@ -1439,7 +1423,8 @@ static int wm8900_probe(struct platform_device *pdev) goto pcm_err; } - wm8900_add_controls(codec); + snd_soc_add_controls(codec, wm8900_snd_controls, + ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde7454..6ff34b9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume", 0, 63, 0, out_tlv), }; -static int wm8903_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8903_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new linput_mode_mux = SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum); @@ -1737,7 +1722,8 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - wm8903_add_controls(socdev->codec); + snd_soc_add_controls(socdev->codec, wm8903_snd_controls, + ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f..c8bd9b0 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), }; -/* add non-DAPM controls */ -static int wm8971_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8971_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -745,7 +730,8 @@ static int wm8971_init(struct snd_soc_device *socdev) reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); - wm8971_add_controls(codec); + snd_soc_add_controls(codec, wm8971_snd_controls, + ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc1..6b27786 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -417,21 +417,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, }; -/* add non dapm controls */ -static int wm8990_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8990_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1460,7 +1445,8 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8990_add_controls(codec); + snd_soc_add_controls(codec, wm8990_snd_controls, + ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d62..1b0ace0 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), }; -/* add non dapm controls */ -static int wm9712_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9712_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. @@ -698,7 +683,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm9712_add_controls(codec); + snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aa..a456226 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -190,21 +190,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -/* add non dapm controls */ -static int wm9713_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9713_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -1245,7 +1230,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - wm9713_add_controls(codec); + snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cbe7e8..3167162 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1495,6 +1495,29 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, EXPORT_SYMBOL_GPL(snd_soc_cnew); /** + * snd_soc_add_controls - add an array of controls to a codec. + * Convienience function to add a list of controls. Many codecs were + * duplicating this code. + * + * Return 0 for success, else error. + */ +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls) +{ + int err, i; + + for (i = 0; i < num_controls; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_add_controls); + +/** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control * @uinfo: control element information -- 1.5.6.5

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Re: [alsa-devel] Bug tracking help wanted (Re: alsa driver #2234 SoundScape Elite not support)
by Krzysztof Helt 08 Jan '09

08 Jan '09

On Wed, 7 Jan 2009 19:13:18 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > Hi Krzysztof, > > Yes, the Soundscape Elite does have the AD1848KP chip. > > I compiled the 2.6.28 kernel and enabled the snd-sscape > module, though I compiled the rest of the ALSA code into > the kernel. It is closer, but not there yet; I get some > noise (clicks) from the card when trying to access it but > no sound. I ran the alsamixer and I also tried to upload > firmware with the sscape_ctl program, but the only message > I get from doing that is "ALSA sound/isa/sscape.c:536: > sscape: No response from on-board processor after upload" > It seems that the codec is detected correctly now so try running alsamixer, set volume to non-zero level (maximum is the best for the first trial but will be too loud probably) then use aplay command to play any sound. Do not upload card's firmware as it maybe needed only for MIDI playback (OSS driver does not need it). Let's try running the digital playback and recording without the midi part first. What is error message produced if you try using aplay? Is there anything new in the dmesg file? I will try comparing the source of the free OSS driver to find out is there any additional actions inside driver required for the Elite card. Regards, Krzysztof ---------------------------------------------------------------------- Znamy wszystkie lekarstwa na nude! Przekonaj sie sam. Kliknij >> http://link.interia.pl/f200d

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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 08 Jan '09

08 Jan '09

At Thu, 8 Jan 2009 12:10:27 -0500, Kyler Laird wrote: > > On Thu, Jan 08, 2009 at 04:52:59PM +0100, Takashi Iwai wrote: > > > > There appears to be no difference in output between headphones plugged > > > and unplugged. > > > > Thanks. So, it implies that the device actually doesn't need > > the headphone detection but rather does on hardware by itself. > > > > How about the patch below? > > It's against the latest git but should work with 2.6.28, too. > > The patch applied alright but did not seem to help. I applied my patch > on top of it and it did work again. That surprised me. It seems that > your patch just disables hp_detect but apparently STAC_HP_EVENT is still > being received by stac92xx_unsol_event() and that's calling > stac92xx_hp_detect(). > > I added a printk() to stac92xx_unsol_event() and see that it's called > only once, right as the module is loaded (res=3221225472). It is not > called again when I plug or unplug the headphones. > > I agree that there does not seem to be headphone detection but it does > seem that another audio pathway needs to be enabled in order for the > headphones to work. Could you add dump_stack() in stac92xx_unsol_event() and check the code path? You are using 2.6.28, not the latest git tree, right? thanks, Takashi

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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 08 Jan '09

08 Jan '09

Hi, sorry for the late reply, as I lately got back from vacation. At Thu, 1 Jan 2009 09:57:30 -0500, Kyler Laird wrote: > > On Thu, Jan 01, 2009 at 11:04:06AM +0100, Takashi Iwai wrote: > > > Please run alsa-info.sh with --no-upload option and attach the > > generated files on both the headphone plugged and unplugged. > > Also, run preferably on both (old) working and non-working kernels > > to compare. > > There appears to be no difference in output between headphones plugged > and unplugged. Thanks. So, it implies that the device actually doesn't need the headphone detection but rather does on hardware by itself. How about the patch below? It's against the latest git but should work with 2.6.28, too. Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 35b83dc..b6040b7 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4408,6 +4408,9 @@ static int patch_stac9200(struct hda_codec *codec) return err; } + if (spec->board_config == STAC_9200_PANASONIC) + spec->hp_detect = 0; + codec->patch_ops = stac92xx_patch_ops; return 0;

1 0

[alsa-devel] [PATCH 3/3] ASoC: TWL4030: Module unloading fix
by Peter Ujfalusi 08 Jan '09

08 Jan '09

Call the snd_soc_free_pcm and snd_soc_dapm_free when the codec driver is unloaded. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index db24f83..ea370a4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1280,6 +1280,8 @@ static int twl4030_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); kfree(codec); return 0; -- 1.6.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Merge the soc_value_enum to soc_enum struct
by Peter Ujfalusi 08 Jan '09

08 Jan '09

Merge the recently introduced soc_value_enum structure to the soc_enum. The value based enums are still handled separately from the normal enum types, but with the merge some of the newly introduced functions can be removed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 15 +-------- sound/soc/soc-core.c | 37 +-------------------- sound/soc/soc-dapm.c | 80 +++------------------------------------------- 4 files changed, 9 insertions(+), 125 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 4af1083..93a4edb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -178,7 +178,7 @@ .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_value_enum_double, \ + .info = snd_soc_info_enum_double, \ .get = snd_soc_dapm_get_value_enum_double, \ .put = snd_soc_dapm_put_value_enum_double, \ .private_value = (unsigned long)&xenum } diff --git a/include/sound/soc.h b/include/sound/soc.h index 9b930d3..24593ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -106,7 +106,7 @@ .private_value = (unsigned long)&xenum } #define SOC_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ - .info = snd_soc_info_value_enum_double, \ + .info = snd_soc_info_enum_double, \ .get = snd_soc_get_value_enum_double, \ .put = snd_soc_put_value_enum_double, \ .private_value = (unsigned long)&xenum } @@ -211,8 +211,6 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_info_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, @@ -419,17 +417,6 @@ struct soc_enum { unsigned char shift_l; unsigned char shift_r; unsigned int max; - const char **texts; - void *dapm; -}; - -/* semi enumerated kcontrol */ -struct soc_value_enum { - unsigned short reg; - unsigned short reg2; - unsigned char shift_l; - unsigned char shift_r; - unsigned int max; unsigned int mask; const char **texts; const unsigned int *values; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cbe7e8..55fdb4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1585,37 +1585,6 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** - * snd_soc_info_value_enum_double - semi enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double semi enumerated - * mixer control. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. - */ -int snd_soc_info_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->max; - - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_value_enum_double); - -/** * snd_soc_get_value_enum_double - semi enumerated double mixer get callback * @kcontrol: mixer control * @ucontrol: control element information @@ -1631,8 +1600,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); @@ -1671,8 +1639,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; unsigned short mask; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ad0d801..493a4e8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -137,7 +137,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } break; case snd_soc_dapm_value_mux: { - struct soc_value_enum *e = (struct soc_value_enum *) + struct soc_enum *e = (struct soc_enum *) w->kcontrols[i].private_value; int val, item; @@ -200,30 +200,6 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, return -ENODEV; } -/* connect value_mux widget to it's interconnecting audio paths */ -static int dapm_connect_value_mux(struct snd_soc_codec *codec, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, - struct snd_soc_dapm_path *path, const char *control_name, - const struct snd_kcontrol_new *kcontrol) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - int i; - - for (i = 0; i < e->max; i++) { - if (!(strcmp(control_name, e->texts[i]))) { - list_add(&path->list, &codec->dapm_paths); - list_add(&path->list_sink, &dest->sources); - list_add(&path->list_source, &src->sinks); - path->name = (char *)e->texts[i]; - dapm_set_path_status(dest, path, 0); - return 0; - } - } - - return -ENODEV; -} - /* connect mixer widget to it's interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, @@ -774,45 +750,6 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } -/* test and update the power status of a value_mux widget */ -static int dapm_value_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_value_enum *e) -{ - struct snd_soc_dapm_path *path; - int found = 0; - - if (widget->id != snd_soc_dapm_value_mux) - return -ENODEV; - - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) - return 0; - - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { - if (path->kcontrol != kcontrol) - continue; - - if (!path->name || !e->texts[mux]) - continue; - - found = 1; - /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) - path->connect = 1; /* new connection */ - else - path->connect = 0; /* old connection must be - powered down */ - } - - if (found) { - dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mux power update"); - } - - return 0; -} - /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int reg, @@ -1045,17 +982,12 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->connect = 1; return 0; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: ret = dapm_connect_mux(codec, wsource, wsink, path, control, &wsink->kcontrols[0]); if (ret != 0) goto err; break; - case snd_soc_dapm_value_mux: - ret = dapm_connect_value_mux(codec, wsource, wsink, path, - control, &wsink->kcontrols[0]); - if (ret != 0) - goto err; - break; case snd_soc_dapm_switch: case snd_soc_dapm_mixer: ret = dapm_connect_mixer(codec, wsource, wsink, path, control); @@ -1382,8 +1314,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); @@ -1423,8 +1354,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, mux; unsigned short mask; int ret = 0; @@ -1443,7 +1373,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_value_mux_update_power(widget, kcontrol, mask, mux, val, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.6.1

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[alsa-devel] Autodetection for Acer Aspire 5930G and 6930G
by Łukasz Wojniłowicz 08 Jan '09

08 Jan '09

This is a patch which adds correct auto detection of model for snd-hda-intel for Acer Aspire 5930G and 6930G. Tested on my 5930G. It finally adds hp jack sense and 5.1 speaker system sliders. Information about 5930G Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x1025013f Revision Id: 0x100202 Information about 6930G Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x1025015e Revision Id: 0x100202 It is first patch prepared by me and it is against commit daead538e9d8efe61d7d2bd12993c8d961b0abd6 from "http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=daead538e9d8efe61…" --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8461,5 +8461,6 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", - ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */

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Re: [alsa-devel] Driver for TI/BB PCM3006 codec
by Mark Brown 08 Jan '09

08 Jan '09

On Thu, Jan 08, 2009 at 11:16:00AM +0100, Davide Rizzo wrote: > This patch implements the driver for the TI (ex Burr-Brown) PCM3006 codec > > Signed-off-by: Davide Rizzo <elpa-rizzo(a)gmail.com> This looks good but needs a refresh against the current ASoC code - there have been some API changes around codec registration for 2.6.29. You should make this a platform driver, then register the codec and DAIs with the ASoC core when that driver is probed. Current code is in the topic/asoc branc of: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git A few other comments below. > +++ linux-2.6.28.elpa/sound/soc/codecs/Kconfig 2009-01-04 > 08:58:40.000000000 +0100 > @@ -60,6 +60,10 @@ config SND_SOC_CS4270_VD33_ERRATA > bool > depends on SND_SOC_CS4270 > > +config SND_SOC_PCM3006 > + tristate > + depends on SND_SOC > + The depends isn't needed and this should also be added to SND_SOC_ALL_CODECS. > +/* > + * pcm3006.c -- ALSA Soc PCM3006 codec support > + * > + * Revision history > + * 25th Apr 2008 Initial version. Please drop the changelog from the code - git will keep track of this. > +static unsigned int pcm3006_read(struct snd_soc_codec *codec, > + unsigned int reg) > +{ > + return 0; > +} > + > +static int pcm3006_write(struct snd_soc_codec *codec, unsigned int reg, > + unsigned int val) > +{ > + return 0; > +} Just remove these if the codec doesn't have any registers, they are only needed if that is the case. > 12:30:42.000000000 +0200 > @@ -0,0 +1,19 @@ > +/* > + * linux/sound/codecs/i2s.h -- ALSA SoC Layer > + * Looks like bitrot here...

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[alsa-devel] 1/7 unstable does not compile: error: ‘SW_VIDEOOUT_INSERT’ undeclared here (not in a function)
by Susan Cragin 08 Jan '09

08 Jan '09

Here is last few lines of output. make[1]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable' make -C /lib/modules/2.6.28-4-generic/build SUBDIRS=/home/susan/Desktop/alsa-driver-unstable CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux-headers-2.6.28-4-generic' CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/hrtimer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/hwdep.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memory_wrapper.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memalloc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sgbuf.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_native.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_lib.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_timer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_misc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_memory.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/rawmidi.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/timer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/wrappers.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/misc_driver.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sound.o /home/susan/Desktop/alsa-driver-unstable/acore/sound.c: In function ‘snd_request_other’: /home/susan/Desktop/alsa-driver-unstable/acore/sound.c:99: warning: format not a string literal and no format arguments CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/init.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memory.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/info.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/control.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/misc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/device.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/isadma.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sound_oss.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/info_oss.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/vmaster.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/jack.o In file included from /home/susan/Desktop/alsa-driver-unstable/acore/jack.c:13: /home/susan/Desktop/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:31: error: ‘SW_VIDEOOUT_INSERT’ undeclared here (not in a function) make[3]: *** [/home/susan/Desktop/alsa-driver-unstable/acore/jack.o] Error 1 make[2]: *** [/home/susan/Desktop/alsa-driver-unstable/acore] Error 2 make[1]: *** [_module_/home/susan/Desktop/alsa-driver-unstable] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.28-4-generic' make: *** [compile] Error 2 susan@ubuntu:~/Desktop/alsa-driver-unstable$

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[alsa-devel] [PATCH 0/2] ALSA jack updates for 2.6.30
by Mark Brown 08 Jan '09

08 Jan '09

The following changes since commit 6a94cb73064c952255336cc57731904174b2c58f: Linus Torvalds (1): Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git topic/jack-core Jani Nikula (1): ALSA: Add support for video out to the jack reporting API Mark Brown (1): ALSA: Reduce boilerplate for new jack types include/linux/input.h | 1 + include/sound/jack.h | 5 +++++ sound/core/jack.c | 45 +++++++++++++++++++++------------------------ 3 files changed, 27 insertions(+), 24 deletions(-)

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Re: [alsa-devel] Bug tracking help wanted (Re: alsa driver #2234 SoundScape Elite not support)
by Krzysztof Helt 07 Jan '09

07 Jan '09

On Tue, 6 Jan 2009 13:41:35 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > sscape: hardware detected at 0x330, using IRQ 5, DMA 1 > ALSA sound/isa/cs423x/cs4231_lib.c:226: codec out - reg 0xc = 0x40 > ALSA sound/isa/cs423x/cs4231_lib.c:1038: cs4231: port = 0x534, id = 0xa > ALSA sound/isa/cs423x/cs4231_lib.c:1044: CS4231: VERSION (I25) = 0x0 > ALSA sound/isa/cs423x/cs4231_lib.c:1063: unknown CS chip with version 0x0 Hi, There is one problem here. The codec is not detected. The codec on supported SoundScape boards is AD1845 which is compatible (more or less) with the CS4231 codec (hence cs4231 library). According to the wikipedia, the SS Elite uses AD1848KP codec which is older and not supported by the cs4231 library. Could you confirm that your card had AD1848KP codec (big chip with "Analog Devices" and "SoundPort" captions)? The simplest solution is to try the 2.6.28 or later kernel as the support for CS4231 and AD1848 codecs were unified. The new kernel may just work for you out of the box due to this change. If it does not please send the dmesg output again. Thank for your findings, Krzysztof ---------------------------------------------------------------------- Szef przynudza? Zagraj sobie! Kliknij >> http://link.interia.pl/f200e

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[alsa-devel] [PATCH] ASoC: add new DAPM control
by Ian Molton 07 Jan '09

07 Jan '09

Hi. The attached patch adds a new mixer control that allows cleaner support for the wm9705 codec Im working on. I think it probably has wider uses though. It saves a few bytes too in shorter names :-) If you prefer I have it available in: git://git.mnementh.co.uk/linux-2.6-im.git asoc and on gitweb at: http://git.mnementh.co.uk/cgi-bin/gitweb.cgi?p=linux-2.6-im.git;a=shortlog;…

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Re: [alsa-devel] Asus w5f. ALC660-VD codec not detected
by Takashi Iwai 07 Jan '09

07 Jan '09

At Wed, 7 Jan 2009 12:57:57 +0100, Salvatore Filippone wrote: > > On Wed, Jan 7, 2009 at 12:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 29 Dec 2008 19:45:25 +0100, > Nigel Henry wrote: > > > > UPDATE: Salvatore has found a workaround, so that both codecs are > detected > > correctly. Any help from alsa developers for a permanent fix would be > > appreciated. See workaround at bottom of post. > > > (snip) > > <Quoting Salvatore> > > Saturday 16:05:58 > > > > Ok, I finally found a workaround. > > According to the thread starting at > > http://thread.gmane.org/gmane.linux.alsa.devel/58849 > > I grabbed the latest source code > alsa-driver-1.0.18a.17.g27a32.139.g6e583/ > > compiled with --with-debug=verbose and got the message > > Dec 27 15:13:06 localhost kernel: ALSA > > /mnt/travel/ALSA/alsa-driver-1.0.18a.17.g27a32.139.g6e583/pci/hda/../../ > alsa-kernel/pci/hda/hda_intel.c:779: > > codec_mask = 0x2 > > > > I then figure out that the sound device was at codec#0 when it worked, > > therefore I changed snd_hda_intel.c by forcing codec_mask > > if (!chip->codec_mask) { > > chip->codec_mask = azx_readw(chip, STATESTS); > > chip->codec_mask=0x03; > > snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); > > } > > > > And now the sound works correctly. > > Of course this opens the question of why azx_readw does not return a > > correct mask; > > Well, this implies a BIOS bug. This value is set by BIOS to indicate > which codec slot is available. > > > maybe the developers will be interested and/or will suggest a > > fix that is a little more elegant/widely applicable? And why did the > base > > code work just once, for no clear reason ? (I did not have debug=verbose > at > > that time, so I don't have detailed messages in the system log). > > Not sure about this as I didn't track this bug. > When did it work once? Is the device enabled by BIOS properly? > > To summarize: > > 1. It always did work with FC7. I guess it worked likely just casually. The old driver didn't clear the register bit before probing, so it got uninitialized bits. Takashi

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[alsa-devel] [PATCH 0/1] ALSA: Add support for video out to the jack reporting API
by Jani Nikula 07 Jan '09

07 Jan '09

The following patch is generated against topic/jack-core branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Jani.

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[alsa-devel] How to handle multiple sound cards?
by cocala 07 Jan '09

07 Jan '09

Hi, Our platform has multiple sound cards: Stereo, Surround51, spdif and Bluetooth PCM. These sound cards are registered in random order as some sound card are registered as module or not registered sometimes. So the application can't use directly use the card number for specific sound card. If using alias, the asound.conf also should be changed according to the registered sound card after boot up. What's the best way to handle multiple sound cards?

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[alsa-devel] ice1712 and jack
by mike start 07 Jan '09

07 Jan '09

I've been having difficulty getting an ews88MT (ice1712 chipset) to work in full duplex using the jack audio server (jackd). Capture only mode works fine, even down to 64 samples latency, Playback only mode works fine down to 64 samples latency. Duplex mode causes 'xruns' at even 2048 samples latency. Looking at the interrupt handler, if I add some diagnostics to tell me the difference between the capture and playback DMA counters timing out I see what appears to be clock drift between the playback and capture halves of the ice1712. I also added some debug messages to show what sample rate the alsa driver was setting the hardware to and it agrees with the rate set in my jackd invocation (jackd -R -d alsa -n 2 -p 1024 - r 48000 -d hw:0,0) Also adding diagnostics to jackd to show the time spent in poll waiting for the capture and playback interrupts shows that they start in sync and gradually drift until an 'xrun' occurs when they are '-p' samples apart. According to the ice1712 data sheet the 'professional' multitrack part of this chip (the part my soundcard is using) runs its inputs and outputs from the same master clock so I'm puzzled as to why they are drifting. Does anybody have any more information relating to the ews88mt? - I think the drivers are working fine but the problems are being caused by the capture and playback not being quite in sync which renders it almost useless for professional recording. Anyone have any suggestions?

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[alsa-devel] Asus w5f. ALC660-VD codec not detected
by Nigel Henry 07 Jan '09

07 Jan '09

The person I've been trying to help has fedora Core 7 on this Asus w5f laptop. FC7 has sounds. there are 2 codecs detected, one for the soundcard, and one for the sounds on the onboard modem. Link below for output from alsa-info.sh, and a bit of pasted output. http://www.alsa-project.org/db/?f=93964b2ec68d1f912509bb1ac2ee02a5163ec4a1 !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Fri Dec 12 11:07:58 CET 2008 !!Linux Distribution !!------------------ Fedora release 7 (Moonshine) Fedora release 7 (Moonshine) Fedora release 7 (Moonshine) !!Kernel Information !!------------------ Kernel release: 2.6.23.17-88.fc7 Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.14 Library version: 1.0.14a Utilities version: 1.0.14 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfeb38000 irq 16 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1043:1262 !!Modprobe options (Sound related) !!-------------------------------- snd-card-0: index=0 snd-hda-intel: index=0 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel enable : N id : <NULL> index : 0 model : <NULL> position_fix : 0 probe_mask : -1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x10430000 Revision Id: 0x100001 No Modem Function Group found <skip> Codec: Motorola Si3054 Address: 1 Vendor Id: 0x10573055 Subsystem Id: 0x10431316 Revision Id: 0x100700 Modem Function Group: 0x1 Moving on to his other install. He also has Fedora 10 installed on this machine, but the only codec detected is the modem one (Motorola Si3054), and no sounds. Link below for alsa-info.sh output, and output pasted also below. http://www.alsa-project.org/db/?f=552b7c6386f59813a04150545ab79f92d4af2e8a !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Fri Dec 12 11:16:40 CET 2008 !!Linux Distribution !!------------------ Fedora release 10 (Cambridge) Fedora release 10 (Cambridge) Fedora release 10 (Cambridge) Fedora release 10 (Cambridge) !!Kernel Information !!------------------ Kernel release: 2.6.27.5-117.fc10.i686 Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.17 Library version: Utilities version: 1.0.18rc3 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfeb38000 irq 16 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1043:1262 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 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 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 : <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> 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 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Motorola Si3054 Address: 1 Vendor Id: 0x10573055 Subsystem Id: 0x10431316 Revision Id: 0x100700 Modem Function Group: 0x1 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root root 116, 7 2008-12-12 11:13 /dev/snd/controlC0 crw-rw---- 1 root root 116, 6 2008-12-12 11:13 /dev/snd/hwC0D1 crw-rw---- 1 root root 116, 5 2008-12-12 11:13 /dev/snd/pcmC0D6c crw-rw---- 1 root root 116, 4 2008-12-12 11:13 /dev/snd/pcmC0D6p crw-rw---- 1 root root 116, 3 2008-12-12 11:13 /dev/snd/seq crw-rw---- 1 root root 116, 2 2008-12-12 11:13 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # # Place your global alsa-lib configuration here... # @hooks [ { func load files [ "/etc/alsa/pulse-default.conf" ] errors false } ] !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfeb38000 irq 16' Mixer name : 'Motorola Si3054' Components : 'HDA:10573055' Controls : 2 Simple ctrls : 2 Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module fuse i915 drm i2c_core rfcomm sco bridge stp bnep l2cap sunrpc ip6t_REJECT nf_conntrack_ipv6 ip6table_filter ip6_tables ipv6 cpufreq_ondemand acpi_cpufreq dm_multipath uinput snd_hda_intel snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd_hwdep btusb snd bluetooth firewire_ohci 8139cp firewire_core arc4 ecb crypto_blkcipher crc_itu_t sdhci_pci sdhci mmc_core 8139too gspca_vc032x soundcore iwl3945 mii rfkill iTCO_wdt iTCO_vendor_support gspca_main videodev v4l1_compat usb_storage serio_raw mac80211 cfg80211 joydev video output asus_laptop pcspkr ata_generic pata_acpi I also suggested upgrading to the latest snapshot of the alsa driver, which he did, but no change, and only the modem codec is detected on his Fedora 10 install Any suggestions are really welcome to help this guy Nigel.

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[alsa-devel] What exactly is module snd-hdsnd-hda-intel
by Nigel Henry 07 Jan '09

07 Jan '09

I see someone that had sounds working on a Thinkpad T61, but sounds have recently stopped working. The module snd-hdsnd-hda-intel is listed on his lsmod output as below. I've never seen such a module before. Any info on what this module is? lsmod | grep snd_hda_intel returns: snd_hdsnd_hda_intel 324248 2 snd_pcm 62596 2 snd_pcm_oss,snd_hda_intel snd 45604 12 snd_pcm_oss,snd_mixer_oss,snd_hda_intel,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device snd_page_alloc 7816 2 snd_hda_intel,snd_pcm a_intel 324248 2 snd_pcm 62596 2 snd_pcm_oss,snd_hda_intel snd 45604 12 snd_pcm_oss,snd_mixer_oss,snd_hda_intel,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device snd_page_alloc 7816 2 snd_hda_intel,snd_pcm This is on a Debian Lenny install, using kernel 2.6.26-1-686, and alsa driver version 1.0.16. Codec is AD1984. His soundcard is detected ok, as below. $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfe020000 irq 17 This is only out of academic interest, but as modinfo shows no such module, I'm really puzzled how a non existing module can be loaded. Nigel.

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[alsa-devel] alsa-lib-1.0.18 doesn't build with option '--with-softfloat'
by Cliff Cai 07 Jan '09

07 Jan '09

Hi all, I'm trying to build the latest release alsa-lib-1.0.18 on Blackfin with option '--with-softfloat', but it failed with message: libtool: install: warning: remember to run `libtool --finish /usr/lib/alsa-lib/smixer' ../src/.libs/libasound.so: undefined reference to `_sqrtf' ../src/.libs/libasound.so: undefined reference to `_exp' ../src/.libs/libasound.so: undefined reference to `_log' ../src/.libs/libasound.so: undefined reference to `_sqrt' ../src/.libs/libasound.so: undefined reference to `_fmod' ../src/.libs/libasound.so: undefined reference to `_floor' Since I've selected --with-softfloat, '-lm' should be excluded from ALSA_DEPLIBS. but some source files still depend on libm, such as .../pcm/pcm_ladspa.c;.../mixer/ simple_none.c .../alisp/alisp.c.These there files cause undefined reference as above. Best Regards Cliff

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[alsa-devel] [patch] nvidia hdmi audio disabled with xorg "useEdid" option
by Brian Hinz 07 Jan '09

07 Jan '09

Hi, Attached is a patch to add the IDs necessary to enable the hdmi out on an Asus P5N-EM HDMI motherboard. On a related note though, I'm trying to get this board set up as a mythtv and having some troubles getting both audio & video to work properly with my Soyo 32" LCD TV. This is probably better directed to the nvidia forums, but since I'm submitting a patch anyway I figured I'll put the question out there just in case someone knows... The issue I'm having is that the LCD TV has a huge amount of overscan in all modes, unless I direct the nVidia xorg driver to ignore the EDID, in which case the resolution is perfect but then the audio is disabled! Obviously the EDID provides the TV's audio capabilities to the X driver, and so it stands to reason that in discarding this information the audio might not work, but is there any reason to think that this could be corrected in the alsa driver somehow or does the problem lie entirely in the nvidia binary X driver? Any thoughts/suggestions would be appreciated! Thanks, -brian

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[alsa-devel] [PATCH] alsa-info.sh - added extra checks - FIXED
by Travis Place 07 Jan '09

07 Jan '09

> Since this patch hasnt been applied yet, i made a few small edits to > it, so ill resubmit now.. > > Same Changelog stands.. > > > > > > On 29/12/2008, at 2:25 AM, Travis Place wrote: > >> Changelog: >> >> Added checks for Pulseaudio/Esound/aRts 'sound servers', Check if >> they are 1) installed 2) running >> A few small changes for the locale, and timezone - keep things more >> uniform. >> Bumped version to 0.4.53 >> >> Signed-off-by: Travis Place <wishie(a)wishie.net> >> >> >> <alsa-info-0.4.53.patch> >> >> >> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] Fixing the AICA driver
by Adrian McMenamin 07 Jan '09

07 Jan '09

It's about a year since I last mentioned this so I thought I'd give it another go... But the AICA sound driver I wrote for the Dreamcast still won't work properly with a pure ALSA setup and I don't know why. It works when OSS emulation is on so it appears to be some sort of interaction with the ALSA lib layer. The snd-summy driver also appears to work well with a 'pure' ALSA setup, but when (for instance) I try to replay an MP3 track without OSS emulation on I get this: Playing MPEG stream from ms.mp3 ... MPEG 1.0, Layer: III, Freq: 44100, mode: Joint-Stereo, modext: 2, BPF : 2560 Channels: 2, copyright: No, original: Yes, CRC: No, emphasis: 0. Bitrate: 128 Kbits/s, Extension value: 2 Audio: 1:1 conversion, rate: 44100, encoding: signed 16 bit, channels: 2 ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' ALSA snd_pcm_hw_params_set_access error: Invalid argument Can't find a suitable libao driver. (Is device in use?) Any clues anyone? Best wishes for 2009 and thanks for all help, Adrian

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[alsa-devel] [PATCH] Typo kills AD1882 mixers
by Clemens Fruhwirth 07 Jan '09

07 Jan '09

Typo bug. --- linux-2.6.28-gentoo/sound/pci/hda/patch_analog.c 2009-01-04 10:58:53.000000000 +0100 +++ linux-2.6.28-gentoo.new/sound/pci/hda/patch_analog.c 2009-01-03 21:31:14.000000000 +0100 @@ -4264,13 +4264,13 @@ spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); spec->adc_nids = ad1882_adc_nids; spec->capsrc_nids = ad1882_capsrc_nids; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->input_mux = &ad1882_capture_source; else spec->input_mux = &ad1882a_capture_source; spec->num_mixers = 2; spec->mixers[0] = ad1882_base_mixers; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->mixers[1] = ad1882_loopback_mixers; else spec->mixers[1] = ad1882a_loopback_mixers; -- Fruhwirth Clemens http://clemens.endorphin.org

2 1

[alsa-devel] [PATCH] pcm_ladspa: add support for softfloat
by Mike Frysinger 07 Jan '09

07 Jan '09

The ladspa module uses math functions unconditionally which breaks when building with --with-softfloat. Tweak the code to only call the math functions when HAVE_SOFT_FLOAT is not defined. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- src/pcm/pcm_ladspa.c | 23 ++++++++++++++--------- 1 files changed, 14 insertions(+), 9 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 5161820..0e9321c 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -34,7 +34,9 @@ #include <dirent.h> #include <locale.h> +#ifndef HAVE_SOFT_FLOAT #include <math.h> +#endif #include "pcm_local.h" #include "pcm_plugin.h" @@ -511,36 +513,39 @@ static void snd_pcm_ladspa_get_default_cvalue(const LADSPA_Descriptor * desc, un *val = desc->PortRangeHints[port].LowerBound; break; case LADSPA_HINT_DEFAULT_LOW: - if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) { +#ifndef HAVE_SOFT_FLOAT + if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) *val = exp(log(desc->PortRangeHints[port].LowerBound) * 0.75 + log(desc->PortRangeHints[port].UpperBound) * 0.25); - } else { + else +#endif *val = (desc->PortRangeHints[port].LowerBound * 0.75) + (desc->PortRangeHints[port].UpperBound * 0.25); - } break; case LADSPA_HINT_DEFAULT_MIDDLE: - if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) { +#ifndef HAVE_SOFT_FLOAT + if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) *val = sqrt(desc->PortRangeHints[port].LowerBound * desc->PortRangeHints[port].UpperBound); - } else { + else +#endif *val = 0.5 * (desc->PortRangeHints[port].LowerBound + desc->PortRangeHints[port].UpperBound); - } break; case LADSPA_HINT_DEFAULT_HIGH: - if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) { +#ifndef HAVE_SOFT_FLOAT + if (LADSPA_IS_HINT_LOGARITHMIC(hdesc)) *val = exp(log(desc->PortRangeHints[port].LowerBound) * 0.25 + log(desc->PortRangeHints[port].UpperBound) * 0.75); - } else { + else +#endif *val = (desc->PortRangeHints[port].LowerBound * 0.25) + (desc->PortRangeHints[port].UpperBound * 0.75); - } break; case LADSPA_HINT_DEFAULT_MAXIMUM: *val = desc->PortRangeHints[port].UpperBound; -- 1.6.0.6

2 2

[alsa-devel] alsa-info security/privacy concerns
by Wu Fengguang 07 Jan '09

07 Jan '09

Greetings, The auto-uploading behavior of http://www.alsa-project.org/alsa-info.sh makes it act like a spyware. This makes some users - including me in the first try - astonishing and uncomfortable. I'd like to fix it by 1) adding user prompt 2) defaulting to --no-upload and introduce --upload Before going for the code, I'd like to ask for your opinions, especially on (2). Thank you. Fengguang

3 4

[alsa-devel] [PATCH] Add config files for more Analog Devices codecs
by Mike Frysinger 07 Jan '09

07 Jan '09

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- src/conf/cards/AD1836.conf | 17 +++++++++++++++++ src/conf/cards/AD1980.conf | 42 ++++++++++++++++++++++++++++++++++++++++++ src/conf/cards/AD73311.conf | 14 ++++++++++++++ src/conf/cards/AD73322.conf | 17 +++++++++++++++++ src/conf/cards/Makefile.am | 5 +++++ src/conf/cards/SSM2602.conf | 12 ++++++++++++ 6 files changed, 107 insertions(+), 0 deletions(-) create mode 100644 src/conf/cards/AD1836.conf create mode 100644 src/conf/cards/AD1980.conf create mode 100644 src/conf/cards/AD73311.conf create mode 100644 src/conf/cards/AD73322.conf create mode 100644 src/conf/cards/SSM2602.conf diff --git a/src/conf/cards/AD1836.conf b/src/conf/cards/AD1836.conf new file mode 100644 index 0000000..0317ec5 --- /dev/null +++ b/src/conf/cards/AD1836.conf @@ -0,0 +1,17 @@ +AD1836.pcm.default { + type plug + slave { + pcm { + type mmap_emul + slave { + pcm { + type hw + card 0 + format S32_LE + rate 48000 + sync_ptr_ioctl 1 + } + } + } + } +} diff --git a/src/conf/cards/AD1980.conf b/src/conf/cards/AD1980.conf new file mode 100644 index 0000000..bd0e519 --- /dev/null +++ b/src/conf/cards/AD1980.conf @@ -0,0 +1,42 @@ +<confdir:pcm/front.conf> + +AD1980.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 +} + +AD1980.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:CARD=" $CARD ] + } + } + capture.pcm { + type plug + slave.pcm { + @func concat + strings [ "dsnoop:CARD=" $CARD ] + } + } +} + +<confdir:pcm/surround40.conf> +<confdir:pcm/surround41.conf> +<confdir:pcm/surround50.conf> +<confdir:pcm/surround51.conf> + +AD1980.pcm.surround40.0 cards.AD1980.pcm.front.0 +AD1980.pcm.surround51.0 cards.AD1980.pcm.front.0 + diff --git a/src/conf/cards/AD73311.conf b/src/conf/cards/AD73311.conf new file mode 100644 index 0000000..e271e9c --- /dev/null +++ b/src/conf/cards/AD73311.conf @@ -0,0 +1,14 @@ +AD73311.pcm.default { + type plug + slave { + pcm { + type hw + card 0 + format S16_LE + rate 8000 + channels 1 + } + + + } +} diff --git a/src/conf/cards/AD73322.conf b/src/conf/cards/AD73322.conf new file mode 100644 index 0000000..e6cbc29 --- /dev/null +++ b/src/conf/cards/AD73322.conf @@ -0,0 +1,17 @@ +AD73322.pcm.default { + type plug + slave { + pcm { + type mmap_emul + slave { + pcm { + type hw + card 0 + format S16_LE + rate 8000 + channels 1 + } + } + } + } +} diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index f4d6c17..f2d34c7 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -2,6 +2,10 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ alsadir = $(alsaconfigdir)/cards cfg_files = aliases.conf \ AACI.conf \ + AD1836.conf \ + AD1980.conf \ + AD73311.conf \ + AD73322.conf \ ATIIXP.conf \ ATIIXP-SPDMA.conf \ ATIIXP-MODEM.conf \ @@ -41,6 +45,7 @@ cfg_files = aliases.conf \ RME9636.conf \ RME9652.conf \ SI7018.conf \ + SSM2602.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ diff --git a/src/conf/cards/SSM2602.conf b/src/conf/cards/SSM2602.conf new file mode 100644 index 0000000..a6fbffe --- /dev/null +++ b/src/conf/cards/SSM2602.conf @@ -0,0 +1,12 @@ +SSM2602.pcm.default { + type plug + slave { + pcm { + type hw + card 0 + format S32_LE + rate 48000 + channels 2 + } + } +} -- 1.6.0.6

1 0

Re: [alsa-devel] Bug tracking help wanted (Re: alsa driver #2234 SoundScape Elite not support)
by Krzysztof Helt 06 Jan '09

06 Jan '09

On Wed, 31 Dec 2008 15:51:56 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > As a new Slackware was just released this month, I have been > upgrading and tweaking my new 2.6.27.7 kernel and just noticed this > week that last year there was some activity on the ISA SoundScape non- > PnP ALSA driver. I have played with it some more, but have not > gotten it to work. I'm still interested in getting a working ALSA > driver for the SoundScape Elite, so if a developer has code for me to > test, I will. I have only SoundScape VIVO card, but I can provide some help if you can send me the output from the driver (snd_sscape module). Please compile kernel with sound options: "Verbose printk", "Debug" and "More verbose debug". I suspect the Elite model requires some firmware to be uploaded before the card is usable. Kind regards, Krzysztof Dzwon tanio do wszystkich! Sprawdz >> http://link.interia.pl/f200a

1 0

[alsa-devel] ASoC updates for v2.6.29
by Mark Brown 06 Jan '09

06 Jan '09

The following changes since commit 2f42357722f7ddc1ec0236fa55ad49ea8a7ce4b0: Peter Ujfalusi (1): ASoC: TWL4030: Convert the bitfield enums to VALUE_ENUM type are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (2): ASoC: Use snd_soc_dapm_nc_pin() in at91sam9g20ek ASoC: Fix SND_SOC_ALL_CODECS handling of dual SPI and I2C control buses sound/soc/atmel/sam9g20_wm8731.c | 4 ++-- sound/soc/codecs/Kconfig | 20 +++++++++++++++----- 2 files changed, 17 insertions(+), 7 deletions(-)

2 3

[alsa-devel] [PATCH - SB-0500 1/1] preliminary support for Toshiba SB-0500 (remote works)
by Andrea Borgia 06 Jan '09

06 Jan '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/usb/usbmixer.c b/usb/usbmixer.c index a492461..6490a74 100644 --- a/usb/usbmixer.c +++ b/usb/usbmixer.c @@ -66,6 +66,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; struct usb_mixer_interface { @@ -1706,7 +1707,8 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, break; /* live24ext: 4 = line-in jack */ case 3: /* hp-out jack (may actuate Mute) */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); break; default: @@ -1957,8 +1959,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { - if (i > 1 && /* Live24ext has 2 LEDs only */ - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (i > 1 && /* Live24ext has 2 LEDs only */ + (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); @@ -1995,7 +1998,8 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) jacks = jacks_audigy2nx; - else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) jacks = jacks_live24ext; else return; @@ -2045,7 +2049,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, goto _error; if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) { + mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { struct snd_info_entry *entry; if ((err = snd_audigy2nx_controls_create(mixer)) < 0) diff --git a/usb/usbmixer_maps.c b/usb/usbmixer_maps.c index d755be0..f41214f 100644 --- a/usb/usbmixer_maps.c +++ b/usb/usbmixer_maps.c @@ -285,6 +285,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = live24ext_map, }, { + .id = USB_ID(0x041e, 0x3048), + .map = audigy2nx_map, + .selector_map = audigy2nx_selectors, + }, + { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), .ignore_ctl_error = 1, -- 1.5.4.3

2 3

[alsa-devel] ASoC updates
by Mark Brown 06 Jan '09

06 Jan '09

The following changes since commit cad9d90ec3f6c497dce2b163046436b6ab5cf6b6: Takashi Iwai (1): Merge branch 'fixes' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai David Brownell (1): ASoC: Clocking fixes for davinci-evm.c Dmitry Baryshkov (1): pxa2xx-ac97: switch AC unit to correct state before probing Grazvydas Ignotas (1): ASoC: Mark non-connected TWL4030 pins for pandora Jarkko Nikula (1): ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers Peter Ujfalusi (2): ASoC: New enum type: value_enum ASoC: TWL4030: Convert the bitfield enums to VALUE_ENUM type include/sound/soc-dapm.h | 15 +++ include/sound/soc.h | 30 ++++++ sound/arm/pxa2xx-ac97-lib.c | 25 +++-- sound/soc/codecs/twl4030.c | 118 +++++++++++------------- sound/soc/davinci/davinci-evm.c | 19 +++- sound/soc/omap/Kconfig | 1 + sound/soc/omap/omap3pandora.c | 13 +++ sound/soc/soc-core.c | 107 +++++++++++++++++++++ sound/soc/soc-dapm.c | 197 ++++++++++++++++++++++++++++++++++++++- 9 files changed, 447 insertions(+), 78 deletions(-)

2 3

[alsa-devel] ASoC updates for 2.6.29
by Mark Brown 05 Jan '09

05 Jan '09

The following changes since commit 1d13bdaa9610e8f7e53f514f429d81b766db101c: Takashi Iwai (1): Merge branch 'topic/hda' into for-next are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git fixes David Brownell (1): ASoC: fix davinci-sffsdr buglet sound/soc/davinci/davinci-sffsdr.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-)

2 1

[alsa-devel] [PATCH] Add AC'97 driver for AT91SAM9263ek atmel board.
by Sedji Gaouaou 05 Jan '09

05 Jan '09

Add AC'97 driver for AT91SAM9263ek atmel board. The driver includes the playback and the capture. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/arm/Kconfig | 9 + sound/arm/Makefile | 4 + sound/arm/at91-ac97.c | 912 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/arm/at91-ac97.h | 295 ++++++++++++++++ 4 files changed, 1220 insertions(+), 0 deletions(-) create mode 100644 sound/arm/at91-ac97.c create mode 100644 sound/arm/at91-ac97.h diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de4..bfe12f7 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -50,5 +50,14 @@ config SND_PXA2XX_AC97 Say Y or M if you want to support any AC97 codec attached to the PXA2xx AC97 interface. +config SND_AT91_AC97 + tristate "AC97 Controller driver for SAM926X familly from ATMEL" + depends on SND && ARCH_AT91 + select SND_PCM + select SND_AC97_CODEC + help + Say Y or M if you want to support any AC97 codec attached to + the SAM926X AC97 Controller. + endif # SND_ARM diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de1..73fc74b 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -17,3 +17,7 @@ snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o snd-pxa2xx-ac97-objs := pxa2xx-ac97.o + +obj-$(CONFIG_SND_AT91_AC97) += snd-at91-ac97.o +snd-at91-ac97-objs := at91-ac97.o + diff --git a/sound/arm/at91-ac97.c b/sound/arm/at91-ac97.c new file mode 100644 index 0000000..3dcf742 --- /dev/null +++ b/sound/arm/at91-ac97.c @@ -0,0 +1,912 @@ +/* + * at91-ac97.c -- AC'97 driver for atmel boards. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2006 Atmel + * + * Author: Patrice Vilchez <patrice.vilchez(a)atmel.com> + * Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/ac97_codec.h> +#include <sound/soc.h> + +#include <asm/cacheflush.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/board.h> + +#include "at91-ac97.h" + + +/* + * module parameters + */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for AC97 controller"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for AC97 controller"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable AC97 controller"); + +typedef struct at91_ac97 { + spinlock_t lock; + void *regs; + int period; + struct clk *ac97_clk; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_ac97 *ac97; + struct snd_ac97_bus *ac97_bus; + int irq; + struct platform_device *pdev; + u8 reset_pin; +} at91_ac97_t; + + +/* + * PIO management functions + */ +void at91_ac97c_drive_reset(at91_ac97_t *chip, unsigned int value) +{ + at91_set_gpio_value(chip->reset_pin, value); +} + +static const char driver_name[] = "at91-ac97"; + + +/* + * PCM part + */ +static struct snd_pcm_hardware snd_at91_ac97_playback_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = (SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S16_BE), + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 256*1024, + .period_bytes_min = 1024, + .period_bytes_max = 4*1024, + .periods_min = 1, + .periods_max = 64, +}; + +static struct snd_pcm_hardware snd_at91_ac97_capture_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = (SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S16_BE), + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 256*1024, + .period_bytes_min = 1024, + .period_bytes_max = 4*1024, + .periods_min = 1, + .periods_max = 64, +}; + +static int snd_at91_ac97_playback_open(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_at91_ac97_playback_hw; + chip->playback_substream = substream; + chip->period = 0; + + pr_debug("%s : snd_at91_ac97_playback_open\n\r", driver_name); + + return 0; +} + +static int snd_at91_ac97_capture_open(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_at91_ac97_capture_hw; + chip->capture_substream = substream; + chip->period = 0; + + pr_debug("%s : snd_at91_ac97_capture_open\n\r", driver_name); + + return 0; +} + +static int snd_at91_ac97_playback_close(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + + return 0; +} + +static int snd_at91_ac97_capture_close(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + + return 0; +} + +static int snd_at91_ac97_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + return err; +} + +static int snd_at91_ac97_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + return err; +} + +static int snd_at91_ac97_playback_hw_free(struct snd_pcm_substream *substream) +{ + + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_at91_ac97_capture_hw_free(struct snd_pcm_substream *substream) +{ + + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_at91_ac97_playback_prepare(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); + unsigned long word = 0; + + pr_debug("%s : block_size = %d\n\r", driver_name, block_size); + + /* Assign slots to channels */ + switch (substream->runtime->channels) { + /* TODO: Support more than two channels */ + case 1: + word |= AT91C_AC97C_CHID3_CA; + break; + case 2: + default: + /* Assign Left and Right slots (3,4) to Channel A */ + word |= AT91C_AC97C_CHID3_CA | AT91C_AC97C_CHID4_CA; + break; + } + + ac97c_writel(chip, OCA, word); + + /* + * Configure sample format and size.. + */ + word = AT91C_AC97C_PDCEN | AT91C_AC97C_SIZE_16_BITS; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_BE: + word |= AT91C_AC97C_CEM; + break; + case SNDRV_PCM_FORMAT_S16_LE: + default: + break; + } + + ac97c_writel(chip, CAMR, word); + + /* Set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AT91C_AC97C_VRA; + ac97c_writel(chip, MR, word); + } else { + /* Clear Variable Rate Bit */ + word = ac97c_readl(chip, MR); + word &= ~AT91C_AC97C_VRA; + ac97c_writel(chip, MR, word); + } + + /* Set rate */ + snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate); + + pr_debug("%s : dma_addr = %x\n\r : dma_area = %x\n\r" + " : dma_bytes = %d\n\r", + driver_name, runtime->dma_addr, + runtime->dma_area, runtime->dma_bytes); + + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TCR); + writel(runtime->dma_addr + block_size, chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + + /* Enable Channel A interrupts */ + ac97c_writel(chip, IER, AT91C_AC97C_CAEVT); + + pr_debug("%s : snd_at91_ac97_playback_prepare\n\r", driver_name); + + return 0; +} + +static int snd_at91_ac97_capture_prepare(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); + unsigned long word = 0; + + pr_debug("%s : block_size = %d\n\r", driver_name, block_size); + + + /* Assign slots to channels */ + switch (substream->runtime->channels) { + /* TODO: Support more than two channels */ + case 1: + word |= AT91C_AC97C_CHID3_CA; + break; + case 2: + default: + /* Assign Left and Right slots (3,4) to Channel A */ + word |= AT91C_AC97C_CHID3_CA | AT91C_AC97C_CHID4_CA; + break; + } + + ac97c_writel(chip, ICA, word); + + /* + * Configure sample format and size. + */ + word = AT91C_AC97C_PDCEN | AT91C_AC97C_SIZE_16_BITS; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_BE: + word |= AT91C_AC97C_CEM; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + } + + ac97c_writel(chip, CAMR, word); + + /* Set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AT91C_AC97C_VRA; + ac97c_writel(chip, MR, word); + } else { + /* Clear Variable Rate Bit */ + word = ac97c_readl(chip, MR); + word &= ~AT91C_AC97C_VRA; + ac97c_writel(chip, MR, word); + } + + /* Set rate */ + snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate); + + pr_debug("%s : dma_addr = %x\n\r : dma_area = %x\n\r" + " : dma_bytes = %d\n\r", + driver_name, runtime->dma_addr, + runtime->dma_area, runtime->dma_bytes); + + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RCR); + writel(runtime->dma_addr + block_size, chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + + /* Enable Channel A interrupts */ + ac97c_writel(chip, IER, AT91C_AC97C_CAEVT); + + pr_debug("%s : snd_at91_ac97_capture_prepare\n\r", driver_name); + + return 0; +} + +static int snd_at91_ac97_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + unsigned long camr, ptcr = 0, flags; + int err = 0; + + spin_lock_irqsave(&chip->lock, flags); + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + camr |= (AT91C_AC97C_CEN | AT91C_AC97C_ENDTX); + ptcr = ATMEL_PDC_TXTEN; + break; + case SNDRV_PCM_TRIGGER_STOP: + camr &= ~(AT91C_AC97C_CEN | AT91C_AC97C_ENDTX); + ptcr = ATMEL_PDC_TXTDIS; + break; + default: + err = -EINVAL; + break; + } + + ac97c_writel(chip, CAMR, camr); + + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); + + spin_unlock_irqrestore(&chip->lock, flags); + + pr_debug("%s : snd_at91_ac97_playback_trigger\n\r", driver_name); + + return err; +} + +static int snd_at91_ac97_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + unsigned long camr, ptcr = 0, flags; + int err = 0; + + spin_lock_irqsave(&chip->lock, flags); + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + camr |= (AT91C_AC97C_CEN | + AT91C_AC97C_ENDRX | + AT91C_AC97C_OVRUN); + ptcr = ATMEL_PDC_RXTEN; + break; + case SNDRV_PCM_TRIGGER_STOP: + camr &= ~(AT91C_AC97C_CEN | + AT91C_AC97C_ENDRX | + AT91C_AC97C_OVRUN); + ptcr = ATMEL_PDC_RXTDIS; + break; + default: + err = -EINVAL; + break; + } + + ac97c_writel(chip, CAMR, camr); + + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); + + spin_unlock_irqrestore(&chip->lock, flags); + + pr_debug("%s : snd_at91_ac97_capture_trigger\n\r", driver_name); + + return err; +} + +static snd_pcm_uframes_t +snd_at91_ac97_playback_pointer(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = readl(chip->regs + ATMEL_PDC_TPR) - runtime->dma_addr; + + pos = bytes_to_frames(runtime, bytes); + + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + pr_debug("%s : snd_at91_ac97_playback_pointer\n\r", driver_name); + + return pos; +} + +static snd_pcm_uframes_t +snd_at91_ac97_capture_pointer(struct snd_pcm_substream *substream) +{ + at91_ac97_t *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = readl(chip->regs + ATMEL_PDC_RPR) - runtime->dma_addr; + pos = bytes_to_frames(runtime, bytes); + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + pr_debug("%s : snd_at91_ac97_capture_pointer\n\r", driver_name); + + return pos; +} + +static struct snd_pcm_ops at91_ac97_playback_ops = { + .open = snd_at91_ac97_playback_open, + .close = snd_at91_ac97_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at91_ac97_playback_hw_params, + .hw_free = snd_at91_ac97_playback_hw_free, + .prepare = snd_at91_ac97_playback_prepare, + .trigger = snd_at91_ac97_playback_trigger, + .pointer = snd_at91_ac97_playback_pointer, +}; + +static struct snd_pcm_ops at91_ac97_capture_ops = { + .open = snd_at91_ac97_capture_open, + .close = snd_at91_ac97_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at91_ac97_capture_hw_params, + .hw_free = snd_at91_ac97_capture_hw_free, + .prepare = snd_at91_ac97_capture_prepare, + .trigger = snd_at91_ac97_capture_trigger, + .pointer = snd_at91_ac97_capture_pointer, +}; + +static struct ac97_pcm at91_ac97_pcm_defs[] __devinitdata = { + /* Playback */ + { + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } }, + }, + /* PCM in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } } + }, + /* Mic in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = (1<<AC97_SLOT_MIC), + } } + }, +}; + +static int __devinit snd_at91_ac97_pcm_new(at91_ac97_t *chip) +{ + struct snd_pcm *pcm; + int err; + + err = snd_ac97_pcm_assign(chip->ac97_bus, + ARRAY_SIZE(at91_ac97_pcm_defs), + at91_ac97_pcm_defs); + if (err) + return err; + + err = snd_pcm_new(chip->card, "Atmel AC97", 0, 1, 1, &pcm); + if (err) + return err; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pdev->dev, + 128 * 1024, 256 * 1024); + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &at91_ac97_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &at91_ac97_capture_ops); + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, "Atmel AC97"); + chip->pcm = pcm; + + return 0; +} + + +/* + * Mixer part + */ +static int snd_at91_ac97_mixer_new(at91_ac97_t *chip) +{ + int err; + struct snd_ac97_template template; + + memset(&template, 0, sizeof(template)); + template.private_data = chip; + template.num = 0; + template.addr = 0; + err = snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97); + + return err; +} + +static irqreturn_t snd_at91_ac97_interrupt(int irq, void *dev_id) +{ + at91_ac97_t *chip = dev_id; + unsigned long status; + unsigned long dummy; + + status = ac97c_readl(chip, SR); + + if (status & AT91C_AC97C_CAEVT) { + struct snd_pcm_runtime *runtime; + int offset, next_period, block_size; + unsigned long casr, camr, test; + + casr = ac97c_readl(chip, CASR); + camr = ac97c_readl(chip, CAMR); + test = casr & camr; + + if ((casr & camr) & AT91C_AC97C_ENDTX) { + runtime = chip->playback_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + + snd_pcm_period_elapsed(chip->playback_substream); + } + if ((casr & camr) & AT91C_AC97C_ENDRX) { + runtime = chip->capture_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + snd_pcm_period_elapsed(chip->capture_substream); + } + if ((casr & camr) & AT91C_AC97C_OVRUN) + printk(KERN_INFO " AC97_irq - overrun!\n"); + } else { + printk(KERN_WARNING + "Spurious AC97 interrupt, status = 0x%08lx\n", + status); + } + + dummy = ac97c_readl(chip, SR); + + return IRQ_HANDLED; +} + + +/* + * CODEC part + */ +static void snd_at91_ac97_hard_reset(at91_ac97_t *chip) +{ + /* Enable AC97 Controller.*/ + /* Perform a cold (hard) reset of the AC97 codec.*/ + ac97c_writel(chip, MR, 0); + ac97c_writel(chip, MR, AT91C_AC97C_ENA); + + at91_ac97c_drive_reset(chip, 0); + udelay(1); + at91_ac97c_drive_reset(chip, 1); + udelay(1); +} + +static void snd_at91_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + at91_ac97_t *chip = ac97->private_data; + unsigned long word; + int timeout = 0x100; + + pr_debug("%s : Writing codec register 0x%x = 0x%x\n\r", + driver_name, reg, val); + + word = (reg & 0x7f) << 16 | val; + + do { + if (ac97c_readl(chip, COSR) & AT91C_AC97C_TXRDY) { + ac97c_writel(chip, COTHR, word); + return; + } + udelay(1); + } while (--timeout); + + snd_printk(KERN_WARNING "at91-ac97: codec write timeout\n\r"); +} + +static unsigned short snd_at91_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + at91_ac97_t *chip = ac97->private_data; + unsigned long word; + int timeout = 40; + + word = (0x80 | (reg & 0x7f)) << 16; + + do { + if (ac97c_readl(chip, COSR) & AT91C_AC97C_TXRDY) { + ac97c_writel(chip, COTHR, word); + break; + } + udelay(1); + } while (--timeout); + + if (!timeout) + goto timed_out; + + timeout = 0x100; + + do { + if (ac97c_readl(chip, COSR) & AT91C_AC97C_RXRDY) { + unsigned short val = + (unsigned short) ac97c_readl(chip, CORHR); + return val; + } + udelay(1); + } while (--timeout); + + if (!timeout) + goto timed_out; + +timed_out: + snd_printk(KERN_WARNING "at91-ac97: codec write timeout\n\r"); + return 0xffff; +} + +static void snd_at91_ac97_warm_reset(struct snd_ac97 *ac97) +{ + at91_ac97_t *chip = ac97->private_data; + unsigned int mr = ac97c_readl(chip, MR); + + mr |= AT91C_AC97C_WRST; + + ac97c_writel(chip, MR, mr); + udelay(1); + + mr &= ~AT91C_AC97C_WRST; + ac97c_writel(chip, MR, mr); +} + +static void snd_at91_ac97_destroy(struct snd_card *card) +{ + at91_ac97_t *chip = get_chip(card); + + if (chip->irq != -1) + free_irq(chip->irq, chip); + + if (chip->regs) + iounmap(chip->regs); +} + +static int __devinit snd_at91_ac97_create(struct snd_card *card, + struct platform_device *pdev) +{ + static struct snd_ac97_bus_ops ops = { + .write = snd_at91_ac97_write, + .read = snd_at91_ac97_read, + .reset = snd_at91_ac97_warm_reset, + }; + + at91_ac97_t *chip = get_chip(card); + int irq, err = 0; + + + + card->private_free = snd_at91_ac97_destroy; + + spin_lock_init(&chip->lock); + chip->card = card; + chip->pdev = pdev; + chip->irq = -1; + + if (!(platform_resource_flags(pdev, 0) & IORESOURCE_MEM) + || !(platform_resource_flags(pdev, 1) & IORESOURCE_IRQ)) + return -ENODEV; + + irq = platform_resource_start(pdev, 1); + + err = request_irq(irq, snd_at91_ac97_interrupt, 0, "ac97", chip); + if (err) { + snd_printk(KERN_WARNING "unable to request IRQ%d\n", irq); + return err; + } + + chip->irq = irq; + snd_printk(KERN_INFO "AC97C regs = %08X \n", + platform_resource_start(pdev, 0)); + snd_printk(KERN_INFO "AC97C irq = %d \n", irq); + + chip->regs = ioremap(platform_resource_start(pdev, 0), + platform_resource_len(pdev, 0)); + if (!chip->regs) { + snd_printk(KERN_WARNING "unable to remap AC97C io memory\n"); + return -ENOMEM; + } + + snd_card_set_dev(card, &pdev->dev); + + err = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); + + return err; +} + +static int __devinit snd_at91_ac97_probe(struct platform_device *pdev) +{ + static int dev; + struct atmel_ac97_data *pdata = pdev->dev.platform_data; + struct snd_card *card; + at91_ac97_t *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(at91_ac97_t)); + if (!card) + return -ENOMEM; + chip = get_chip(card); + + err = snd_at91_ac97_create(card, pdev); + if (err) + goto out_free_card; + + /* Enable AC97 Controller clock*/ + chip->reset_pin = pdata->reset_pin; + chip->ac97_clk = clk_get(NULL, "ac97_clk"); + if (!chip->ac97_clk) + goto out_free_card; + + clk_enable(chip->ac97_clk); + + /* Perform a codec hard reset.*/ + /* This also enables the AC97 Controller.*/ + snd_at91_ac97_hard_reset(chip); + + err = snd_at91_ac97_mixer_new(chip); + if (err) + goto out_free_card; + + err = snd_at91_ac97_pcm_new(chip); + if (err) + goto out_free_card; + + strcpy(card->driver, "ac97c"); + strcpy(card->shortname, "Atmel AC97"); + sprintf(card->longname, "Atmel AC97 Controller at %#lx, irq %i", + (unsigned long) platform_resource_start(pdev, 0), + (int) chip->irq); + + err = snd_card_register(card); + if (err) + goto out_free_card; + + dev_set_drvdata(&pdev->dev, card); + dev++; + return 0; + +out_free_card: + snd_card_free(card); + return err; +} + +static int __devexit snd_at91_ac97_remove(struct platform_device *pdev) +{ + struct snd_card *card = dev_get_drvdata(&pdev->dev); + at91_ac97_t *chip = get_chip(card); + + + snd_card_free(card); + + /* Disable AC97 Controller*/ + ac97c_writel(chip, MR, 0); + + /* Disable AC97 Controller clock*/ + clk_disable(chip->ac97_clk); + + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static struct platform_driver at91_ac97_driver = { + .probe = snd_at91_ac97_probe, + .remove = __devexit_p(snd_at91_ac97_remove), + .driver = + { + .name = "ac97c", + } + , +}; + +static int __init at91_ac97_init(void) +{ + return platform_driver_register(&at91_ac97_driver); +} + +static void __exit at91_ac97_exit(void) +{ + platform_driver_unregister(&at91_ac97_driver); +} + +module_init(at91_ac97_init); +module_exit(at91_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel AC97 Controller"); +MODULE_AUTHOR("Atmel"); + diff --git a/sound/arm/at91-ac97.h b/sound/arm/at91-ac97.h new file mode 100644 index 0000000..e945f5d --- /dev/null +++ b/sound/arm/at91-ac97.h @@ -0,0 +1,295 @@ +/* + * at91-ac97.h -- Hardware definition for the ac97c peripheral + * in the ATMEL at91sam926x processor + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2006 Atmel + * + * Author: Patrice Vilchez <patrice.vilchez(a)atmel.com> + * ATMEL CORP. + * + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __AC97C_H +#define __AC97C_H + +/* -------------------------------------------------------- */ +/* AC97C ID definitions for AT91SAM926x */ +/* -------------------------------------------------------- */ +#ifndef AT91C_ID_AC97C +#define AT91C_ID_AC97C 18 /**< AC97 Controller id */ +#endif /* AT91C_ID_AC97C */ + +/* -------------------------------------------------------- */ +/* AC97C Base Address definitions for AT91SAM926x */ +/* -------------------------------------------------------- */ +#define AT91C_BASE_AC97C 0xFFFA0000 /**< AC97C base address */ + +/* -------------------------------------------------------- */ +/* PIO definition for AC97C hardware peripheral */ +/* -------------------------------------------------------- */ +#define AT91C_PB1_AC97CK (1 << 1) /**< */ +#define AT91C_PB0_AC97FS (1 << 0) /**< */ +#define AT91C_PB3_AC97RX (1 << 3) /**< */ +#define AT91C_PB2_AC97TX (1 << 2) /**< */ + +/* -------------------------------------------------------- */ +/* Register offset definition for AC97C hardware peripheral */ +/* -------------------------------------------------------- */ +#define AC97C_MR (0x0008) /**< Mode Register */ +#define AC97C_ICA (0x0010) /**< Input Channel AssignementRegister */ +#define AC97C_OCA (0x0014) /**< Output Channel Assignement Register */ +#define AC97C_CARHR (0x0020) /**< Channel A Receive Holding Register */ +#define AC97C_CATHR (0x0024) /**< Channel A Transmit Holding Register */ +#define AC97C_CASR (0x0028) /**< Channel A Status Register */ +#define AC97C_CAMR (0x002C) /**< Channel A Mode Register */ +#define AC97C_CBRHR (0x0030) /**< Channel B Receive Holding Register (optional) */ +#define AC97C_CBTHR (0x0034) /**< Channel B Transmit Holding Register (optional) */ +#define AC97C_CBSR (0x0038) /**< Channel B Status Register */ +#define AC97C_CBMR (0x003C) /**< Channel B Mode Register */ +#define AC97C_CORHR (0x0040) /**< COdec Transmit Holding Register */ +#define AC97C_COTHR (0x0044) /**< COdec Transmit Holding Register */ +#define AC97C_COSR (0x0048) /**< CODEC Status Register */ +#define AC97C_COMR (0x004C) /**< CODEC Mask Status Register */ +#define AC97C_SR (0x0050) /**< Status Register */ +#define AC97C_IER (0x0054) /**< Interrupt Enable Register */ +#define AC97C_IDR (0x0058) /**< Interrupt Disable Register */ +#define AC97C_IMR (0x005C) /**< Interrupt Mask Register */ +#define AC97C_VERSION (0x00FC) /**< Version Register */ + +/* -------------------------------------------------------- */ +/* Bitfields definition for AC97C hardware peripheral */ +/* -------------------------------------------------------- */ +/* --- Register AC97C_MR */ +#define AT91C_AC97C_ENA (0x1 << 0) /**< (AC97C) AC97 Controller Global Enable */ +#define AT91C_AC97C_WRST (0x1 << 1) /**< (AC97C) Warm Reset */ +#define AT91C_AC97C_VRA (0x1 << 2) /**< (AC97C) Variable RAte (for Data Slots) */ +/* --- Register AC97C_ICA */ +#define AT91C_AC97C_CHID3 (0x7 << 0) /**< (AC97C) Channel Id for the input slot 3 */ +#define AT91C_AC97C_CHID3_NONE 0x0 /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CA 0x1 /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CB 0x2 /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CC 0x3 /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4 (0x7 << 3) /**< (AC97C) Channel Id for the input slot 4 */ +#define AT91C_AC97C_CHID4_NONE (0x0 << 3) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CA (0x1 << 3) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CB (0x2 << 3) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CC (0x3 << 3) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5 (0x7 << 6) /**< (AC97C) Channel Id for the input slot 5 */ +#define AT91C_AC97C_CHID5_NONE (0x0 << 6) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CA (0x1 << 6) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CB (0x2 << 6) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CC (0x3 << 6) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6 (0x7 << 9) /**< (AC97C) Channel Id for the input slot 6 */ +#define AT91C_AC97C_CHID6_NONE (0x0 << 9) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CA (0x1 << 9) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CB (0x2 << 9) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CC (0x3 << 9) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7 (0x7 << 12) /**< (AC97C) Channel Id for the input slot 7 */ +#define AT91C_AC97C_CHID7_NONE (0x0 << 12) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CA (0x1 << 12) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CB (0x2 << 12) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CC (0x3 << 12) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8 (0x7 << 15) /**< (AC97C) Channel Id for the input slot 8 */ +#define AT91C_AC97C_CHID8_NONE (0x0 << 15) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CA (0x1 << 15) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CB (0x2 << 15) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CC (0x3 << 15) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9 (0x7 << 18) /**< (AC97C) Channel Id for the input slot 9 */ +#define AT91C_AC97C_CHID9_NONE (0x0 << 18) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CA (0x1 << 18) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CB (0x2 << 18) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CC (0x3 << 18) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10 (0x7 << 21) /**< (AC97C) Channel Id for the input slot 10 */ +#define AT91C_AC97C_CHID10_NONE (0x0 << 21) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CA (0x1 << 21) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CB (0x2 << 21) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CC (0x3 << 21) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11 (0x7 << 24) /**< (AC97C) Channel Id for the input slot 11 */ +#define AT91C_AC97C_CHID11_NONE (0x0 << 24) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CA (0x1 << 24) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CB (0x2 << 24) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CC (0x3 << 24) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12 (0x7 << 27) /**< (AC97C) Channel Id for the input slot 12 */ +#define AT91C_AC97C_CHID12_NONE (0x0 << 27) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CA (0x1 << 27) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CB (0x2 << 27) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CC (0x3 << 27) /**< (AC97C) Channel C data will be transmitted during this slot */ +/* --- Register AC97C_OCA */ +#define AT91C_AC97C_CHID3 (0x7 << 0) /**< (AC97C) Channel Id for the input slot 3 */ +#define AT91C_AC97C_CHID3_NONE 0x0 /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CA 0x1 /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CB 0x2 /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID3_CC 0x3 /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4 (0x7 << 3) /**< (AC97C) Channel Id for the input slot 4 */ +#define AT91C_AC97C_CHID4_NONE (0x0 << 3) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CA (0x1 << 3) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CB (0x2 << 3) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID4_CC (0x3 << 3) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5 (0x7 << 6) /**< (AC97C) Channel Id for the input slot 5 */ +#define AT91C_AC97C_CHID5_NONE (0x0 << 6) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CA (0x1 << 6) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CB (0x2 << 6) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID5_CC (0x3 << 6) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6 (0x7 << 9) /**< (AC97C) Channel Id for the input slot 6 */ +#define AT91C_AC97C_CHID6_NONE (0x0 << 9) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CA (0x1 << 9) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CB (0x2 << 9) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID6_CC (0x3 << 9) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7 (0x7 << 12) /**< (AC97C) Channel Id for the input slot 7 */ +#define AT91C_AC97C_CHID7_NONE (0x0 << 12) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CA (0x1 << 12) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CB (0x2 << 12) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID7_CC (0x3 << 12) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8 (0x7 << 15) /**< (AC97C) Channel Id for the input slot 8 */ +#define AT91C_AC97C_CHID8_NONE (0x0 << 15) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CA (0x1 << 15) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CB (0x2 << 15) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID8_CC (0x3 << 15) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9 (0x7 << 18) /**< (AC97C) Channel Id for the input slot 9 */ +#define AT91C_AC97C_CHID9_NONE (0x0 << 18) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CA (0x1 << 18) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CB (0x2 << 18) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID9_CC (0x3 << 18) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10 (0x7 << 21) /**< (AC97C) Channel Id for the input slot 10 */ +#define AT91C_AC97C_CHID10_NONE (0x0 << 21) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CA (0x1 << 21) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CB (0x2 << 21) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID10_CC (0x3 << 21) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11 (0x7 << 24) /**< (AC97C) Channel Id for the input slot 11 */ +#define AT91C_AC97C_CHID11_NONE (0x0 << 24) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CA (0x1 << 24) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CB (0x2 << 24) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID11_CC (0x3 << 24) /**< (AC97C) Channel C data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12 (0x7 << 27) /**< (AC97C) Channel Id for the input slot 12 */ +#define AT91C_AC97C_CHID12_NONE (0x0 << 27) /**< (AC97C) No data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CA (0x1 << 27) /**< (AC97C) Channel A data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CB (0x2 << 27) /**< (AC97C) Channel B data will be transmitted during this slot */ +#define AT91C_AC97C_CHID12_CC (0x3 << 27) /**< (AC97C) Channel C data will be transmitted during this slot */ +/* --- Register AC97C_CARHR */ +#define AT91C_AC97C_RDATA (0xFFFFF << 0) /**< (AC97C) Receive data */ +/* --- Register AC97C_CATHR */ +#define AT91C_AC97C_TDATA (0xFFFFF << 0) /**< (AC97C) Transmit data */ +/* --- Register AC97C_CASR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +#define AT91C_AC97C_OVRUN (0x1 << 5) /**< (AC97C) */ +#define AT91C_AC97C_ENDTX (0x1 << 10) /**< (AC97C) */ +#define AT91C_AC97C_TXBUFE (0x1 << 11) /**< (AC97C) */ +#define AT91C_AC97C_ENDRX (0x1 << 14) /**< (AC97C) */ +#define AT91C_AC97C_RXBUFF (0x1 << 15) /**< (AC97C) */ +/* --- Register AC97C_CAMR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +#define AT91C_AC97C_OVRUN (0x1 << 5) /**< (AC97C) */ +#define AT91C_AC97C_ENDTX (0x1 << 10) /**< (AC97C) */ +#define AT91C_AC97C_TXBUFE (0x1 << 11) /**< (AC97C) */ +#define AT91C_AC97C_ENDRX (0x1 << 14) /**< (AC97C) */ +#define AT91C_AC97C_RXBUFF (0x1 << 15) /**< (AC97C) */ +#define AT91C_AC97C_SIZE (0x3 << 16) /**< (AC97C) */ +#define AT91C_AC97C_SIZE_20_BITS (0x0 << 16) /**< (AC97C) Data size is 20 bits */ +#define AT91C_AC97C_SIZE_18_BITS (0x1 << 16) /**< (AC97C) Data size is 18 bits */ +#define AT91C_AC97C_SIZE_16_BITS (0x2 << 16) /**< (AC97C) Data size is 16 bits */ +#define AT91C_AC97C_SIZE_10_BITS (0x3 << 16) /**< (AC97C) Data size is 10 bits */ +#define AT91C_AC97C_CEM (0x1 << 18) /**< (AC97C) */ +#define AT91C_AC97C_CEN (0x1 << 21) /**< (AC97C) */ +#define AT91C_AC97C_PDCEN (0x1 << 22) /**< (AC97C) */ +/* --- Register AC97C_CBRHR */ +#define AT91C_AC97C_RDATA (0xFFFFF << 0) /**< (AC97C) Receive data */ +/* --- Register AC97C_CBTHR */ +#define AT91C_AC97C_TDATA (0xFFFFF << 0) /**< (AC97C) Transmit data */ +/* --- Register AC97C_CBSR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +#define AT91C_AC97C_OVRUN (0x1 << 5) /**< (AC97C) */ +/* --- Register AC97C_CBMR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +#define AT91C_AC97C_OVRUN (0x1 << 5) /**< (AC97C) */ +#define AT91C_AC97C_SIZE (0x3 << 16) /**< (AC97C) */ +#define AT91C_AC97C_SIZE_20_BITS (0x0 << 16) /**< (AC97C) Data size is 20 bits */ +#define AT91C_AC97C_SIZE_18_BITS (0x1 << 16) /**< (AC97C) Data size is 18 bits */ +#define AT91C_AC97C_SIZE_16_BITS (0x2 << 16) /**< (AC97C) Data size is 16 bits */ +#define AT91C_AC97C_SIZE_10_BITS (0x3 << 16) /**< (AC97C) Data size is 10 bits */ +#define AT91C_AC97C_CEM (0x1 << 18) /**< (AC97C) */ +#define AT91C_AC97C_CEN (0x1 << 21) /**< (AC97C) */ +/* --- Register AC97C_CORHR */ +#define AT91C_AC97C_SDATA (0xFFFF << 0) /**< (AC97C) Status Data */ +/* --- Register AC97C_COTHR */ +#define AT91C_AC97C_CDATA (0xFFFF << 0) /**< (AC97C) Command Data */ +#define AT91C_AC97C_CADDR (0x7F << 16) /**< (AC97C) COdec control register index */ +#define AT91C_AC97C_READ (0x1 << 23) /**< (AC97C) Read/Write command */ +/* --- Register AC97C_COSR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +/* --- Register AC97C_COMR */ +#define AT91C_AC97C_TXRDY (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_TXEMPTY (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_UNRUN (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_RXRDY (0x1 << 4) /**< (AC97C) */ +/* --- Register AC97C_SR */ +#define AT91C_AC97C_SOF (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_WKUP (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_COEVT (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_CAEVT (0x1 << 3) /**< (AC97C) */ +#define AT91C_AC97C_CBEVT (0x1 << 4) /**< (AC97C) */ +/* --- Register AC97C_IER */ +#define AT91C_AC97C_SOF (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_WKUP (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_COEVT (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_CAEVT (0x1 << 3) /**< (AC97C) */ +#define AT91C_AC97C_CBEVT (0x1 << 4) /**< (AC97C) */ +/* --- Register AC97C_IDR */ +#define AT91C_AC97C_SOF (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_WKUP (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_COEVT (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_CAEVT (0x1 << 3) /**< (AC97C) */ +#define AT91C_AC97C_CBEVT (0x1 << 4) /**< (AC97C) */ +/* --- Register AC97C_IMR */ +#define AT91C_AC97C_SOF (0x1 << 0) /**< (AC97C) */ +#define AT91C_AC97C_WKUP (0x1 << 1) /**< (AC97C) */ +#define AT91C_AC97C_COEVT (0x1 << 2) /**< (AC97C) */ +#define AT91C_AC97C_CAEVT (0x1 << 3) /**< (AC97C) */ +#define AT91C_AC97C_CBEVT (0x1 << 4) /**< (AC97C) */ + +#define platform_num_resources(dev) ((dev)->num_resources) +#define platform_resource_start(dev, i) ((dev)->resource[(i)].start) +#define platform_resource_end(dev, i) ((dev)->resource[(i)].end) +#define platform_resource_flags(dev, i) ((dev)->resource[(i)].flags) +#define platform_resource_len(dev, i) \ + (platform_resource_end((dev), (i)) - \ + platform_resource_start((dev), (i)) + 1) + +#define get_chip(card) ((at91_ac97_t *)(card)->private_data) + +#define ac97c_writel(chip, reg, val) \ + writel((val), (chip)->regs + AC97C_##reg) + +#define ac97c_readl(chip, reg) \ + readl((chip)->regs + AC97C_##reg) + +#endif /* __AC97C_H */ -- 1.5.3.7

3 7

[alsa-devel] AC'97 driver for AT91SAM9263ek atmel board.
by Sedji Gaouaou 05 Jan '09

05 Jan '09

Hi, I am writing because I have no feedbacks concerning a patch that I have sent on the 1st of october: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011049.ht…. Has anybody tested it? Is there any problem with it? Regards, Sedji

1 0

[alsa-devel] [PATCH 0/2] ASoC: New enum type for bitfield coded enums
by Peter Ujfalusi 05 Jan '09

05 Jan '09

Hello, The first patch adds new enum type to handle non enum coded enums. The twl4030 codec for example have some of it's input/output mux settings in a bitfiled, which can not be handled by the standard enum type. The second patch converts the twl4030 bitfield enums to the new value_enum type. Technically it could be possible to add the new members (mask, *values) to the existing soc_enum structure, which could reduce the amount of new code needed to handle this new type. At least the snd_soc_info_value_enum_double, dapm_connect_value_mux, dapm_value_mux_update_power functions can be removed if I modify the soc_enum instead of introducing the new soc_value_enum. The new type is quite flexible: one can define the normal enum types also with the VALUE_ENUM (although it does not make any sense): /* Normal enum */ static const char *normal_mux_texts[] = {"Off", "mux1", "mux2", "mux3"}; static const unsigned int normal_mux_values[] = {0x0, 0x1, 0x2, 0x3}; static const struct soc_value_enum normal_mux_enum = SOC_VALUE_ENUM_SINGLE(REGISTER, 0, 0x3, ARRAY_SIZE(normal_mux_texts), normal_mux_texts, normal_mux_values); But it makes more sense if one wants to handle the bitfiled (from twl4030 codec): /* PreDrive Right */ static const char *twl4030_predriver_texts[] = {"Off", "DACR1", "DACR2", "DACL2"}; static const unsigned int twl4030_predriver_values[] = {0x0, 0x1, 0x2, 0x4}; static const struct soc_value_enum twl4030_predriver_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, ARRAY_SIZE(twl4030_predriver_texts), twl4030_predriver_texts, twl4030_predriver_values); static const struct snd_kcontrol_new twl4030_dapm_predriver_control = SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);

2 3

[alsa-devel] Bug 3312 -DSP2000 MIDI In
by Alan Horstmann 05 Jan '09

05 Jan '09

This is the 4th bug on the same issue (which is fixed) -see 0055, 1161, 1846. I've added a note; perhaps it could be set resolved or closed? Alan

2 1

[alsa-devel] [PATCH] pxa2xx-ac97: switch AC unit to correct state before probing
by Dmitry Eremin-Solenikov 05 Jan '09

05 Jan '09

From: Dmitry Baryshkov <dbaryshkov(a)gmail.com> If AC97 unit is in partially enabled state, early request_irq can trigger IRQ storm or even full hang up. Workaround this by forcibly switching ACLINK off at the start of the probe. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/arm/pxa2xx-ac97-lib.c | 25 ++++++++++++++++--------- 1 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index ef6539e..35afd0c 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -321,10 +321,6 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); - if (ret < 0) - goto err; - if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -339,7 +335,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); ac97conf_clk = NULL; - goto err_irq; + goto err_conf; } } @@ -347,19 +343,30 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (IS_ERR(ac97_clk)) { ret = PTR_ERR(ac97_clk); ac97_clk = NULL; - goto err_irq; + goto err_clk; } - return clk_enable(ac97_clk); + ret = clk_enable(ac97_clk); + if (ret) + goto err_clk2; + + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + if (ret < 0) + goto err_irq; + + return 0; err_irq: GCR |= GCR_ACLINK_OFF; +err_clk2: + clk_put(ac97_clk); + ac97_clk = NULL; +err_clk: if (ac97conf_clk) { clk_put(ac97conf_clk); ac97conf_clk = NULL; } - free_irq(IRQ_AC97, NULL); -err: +err_conf: return ret; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); -- 1.5.6.5

2 2

[alsa-devel] [patch 2.6.28-git] ASoC: fix davinci-sffsdr buglet
by David Brownell 05 Jan '09

05 Jan '09

From: David Brownell <dbrownell(a)users.sourceforge.net> DaVinci ASoC: fix multiboard buglet Minor bugfix: now that DaVinci kernels can support multiple boards, board-specific ASoC components need to verify they're running on the right board before initializing. Signed-off-by: David Brownell <dbrownell(a)users.sourceforge.net> Signed-off-by: Kevin Hilman <khilman(a)deeprootsystems.com> --- From the DaVinci tree ... didn't realize davinci-sffsdr.c was already queued for mainline. sound/soc/davinci/davinci-sffsdr.c | 4 ++++ 1 file changed, 4 insertions(+) --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -24,6 +24,7 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> +#include <asm/mach-types.h> #include <asm/plat-sffsdr/sffsdr-fpga.h> #include <mach/mcbsp.h> @@ -115,6 +116,9 @@ static int __init sffsdr_init(void) { int ret; + if (!machine_is_sffsdr()) + return -EINVAL; + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); if (!sffsdr_snd_device) { printk(KERN_ERR "platform device allocation failed\n");

2 1

[alsa-devel] [PATCH] ALSA: Add support for video out to the jack reporting API
by Jani Nikula 05 Jan '09

05 Jan '09

Add support for reporting new jack types SND_JACK_VIDEOOUT and SND_JACK_AVOUT (a combination of LINEOUT and VIDEOOUT) to the jack reporting API. Also add the corresponding SW_VIDEOOUT_INSERT switch to the input system header. Signed-off-by: Jani Nikula <ext-jani.1.nikula(a)nokia.com> --- include/linux/input.h | 1 + include/sound/jack.h | 2 ++ sound/core/jack.c | 6 ++++++ 3 files changed, 9 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index 9a6355f..adc1332 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -661,6 +661,7 @@ struct input_absinfo { #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ +#define SW_VIDEOOUT_INSERT 0x08 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 2e0315c..283c975 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -37,6 +37,8 @@ enum snd_jack_types { SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_VIDEOOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12d..e1de195 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -106,6 +106,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_LINEOUT) input_set_capability(jack->input_dev, EV_SW, SW_LINEOUT_INSERT); + if (type & SND_JACK_VIDEOOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_VIDEOOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -163,6 +166,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_LINEOUT) input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, status & SND_JACK_LINEOUT); + if (jack->type & SND_JACK_VIDEOOUT) + input_report_switch(jack->input_dev, SW_VIDEOOUT_INSERT, + status & SND_JACK_VIDEOOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); -- 1.6.0.4

2 1

[alsa-devel] Crackling instead of Audio
by Calad Sigilon 05 Jan '09

05 Jan '09

Got a bit of an interesting problem that #alsa wasn't able to help me solve - though gnubien did get me through a rigorous amount of debugging! No audio output - just crackling. Using position_fix=1 seemed to work for one boot, but then never worked again. Tried position_fix=1, position_fix=2, mode=vaio, no mode=vaio. No effect. Tried pci=nolapic, pci=noacpi, pci=irqpoll, pci=routeirq on the grub boot line. No effect. Firewire is the only other thing on interrupt 21, and I don't use it. Tried blacklisting all modules so HDA Intel was the only thing on interrupt 21 - no effect. It is filed as Bug #4311, which has the alsa-info attached (https://bugtrack.alsa-project.org/alsa-bug/file_download.php?file_id=2818&t…) -- CaladSigilon

1 0

[alsa-devel] added support for remote in Toshiba SB0500
by Andrea Borgia 05 Jan '09

05 Jan '09

Hello. I have a Toshiba SB-0500 (041e:3048), which is basically a USB hub plus ethernet card plus sound card with remote. As of alsa 1.0.18a / linux 2.6.28 this beast is only partially supported by snd-usb-audio and, most importantly for me, the remote does not work (hub and ethernet are ok). Considering the similarities with the SB 24 Live External and with the Audigy, I tried patching usbmixer.c and usbmixer_maps.c until I got it working with lirc's alsa_usb module. Please note that I didn't test mixer functionality and I have not much interest in that part for my use. What should I do to get my patches reviewed and possibly incorporated into the next version? The changes are rather trivial imho but still this is my first serious attempt at kernel hacking. Thanks for your help, Andrea.

1 0

Re: [alsa-devel] [PATCH 2/2] sound/usb: Use negated usb_endpoint_xfer_control, etc
by Takashi Iwai 04 Jan '09

04 Jan '09

At Sat, 3 Jan 2009 17:54:53 +0100 (CET), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > This patch extends 42a6e66f1e40a930d093c33ba0bb9d8d8e4555ed by using > usb_endpoint_xfer_control, usb_endpoint_xfer_isoc, usb_endpoint_xfer_bulk, > and usb_endpoint_xfer_int in the negated case as well. > > This patch also rewrites some calls to usb_endpoint_dir_in as negated calls > to !usb_endpoint_dir_out, and vice versa, to better correspond to the > intent of the original code. > > The semantic patch that makes this change is as follows: > (http://www.emn.fr/x-info/coccinelle/) > > // <smpl> > @@ struct usb_endpoint_descriptor *epd; @@ > > - (usb_endpoint_type(epd) != $USB_ENDPOINT_XFER_CONTROL\|0$) > + !usb_endpoint_xfer_control(epd) > > @@ struct usb_endpoint_descriptor *epd; @@ > > - (usb_endpoint_type(epd) != $USB_ENDPOINT_XFER_ISOC\|1$) > + !usb_endpoint_xfer_isoc(epd) > > @@ struct usb_endpoint_descriptor *epd; @@ > > - (usb_endpoint_type(epd) != $USB_ENDPOINT_XFER_BULK\|2$) > + !usb_endpoint_xfer_bulk(epd) > > @@ struct usb_endpoint_descriptor *epd; @@ > > - (usb_endpoint_type(epd) != $USB_ENDPOINT_XFER_INT\|3$) > + !usb_endpoint_xfer_int(epd) > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Applied now. Thanks. Takashi

1 0

[alsa-devel] [PATCH 2/2] New Control Plugin - arcam-av
by Peter Stokes 03 Jan '09

03 Jan '09

Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk> diff -Nrup alsa-plugins-1.0.18-orig/arcam-av/arcam_av.c alsa-plugins-1.0.18/arcam-av/arcam_av.c --- alsa-plugins-1.0.18-orig/arcam-av/arcam_av.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-plugins-1.0.18/arcam-av/arcam_av.c 2009-01-02 22:30:55.000000000 +0000 @@ -0,0 +1,638 @@ +/* + * ALSA -> Arcam AV control plugin + * + * Copyright (c) 2009 by Peter Stokes <linux(a)dadeos.co.uk> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +#include "arcam_av.h" + +#include <errno.h> +#include <fcntl.h> +#include <semaphore.h> +#include <signal.h> +#include <stddef.h> +#include <stdio.h> +#include <termios.h> +#include <unistd.h> + +#include <sys/ipc.h> +#include <sys/shm.h> +#include <sys/stat.h> +#include <sys/stat.h> +#include <sys/socket.h> +#include <sys/un.h> + +#define MIN(a, b) ((a) < (b) ? (a) : (b)) +#define MAX(a, b) ((a) > (b) ? (a) : (b)) + + +int arcam_av_connect(const char* port) +{ + int fd = open(port, O_RDWR | O_NOCTTY); + if (fd < 0) + return -errno; + + struct termios portsettings; + bzero(&portsettings, sizeof(portsettings)); + portsettings.c_cflag = B38400 | CS8 | CLOCAL | CREAD; + portsettings.c_iflag = IGNPAR; + portsettings.c_oflag = 0; + portsettings.c_lflag = 0; + portsettings.c_cc[VTIME] = 0; + portsettings.c_cc[VMIN] = 5; + tcflush(fd, TCIFLUSH); + tcsetattr(fd, TCSANOW, &portsettings); + + return fd; +} + + +int arcam_av_send(int fd, arcam_av_cc_t command, unsigned char param1, unsigned char param2) +{ + char buffer[7] = {'P', 'C', '_', command, param1, param2, 0x0D}; + + tcdrain(fd); + ssize_t bytes = write(fd, buffer, 7); + + if (bytes == 7) + return 0; + else if (bytes >= 0) + return -1; + else + return -errno; +} + + +static int arcam_av_receive(int fd, arcam_av_cc_t* command, unsigned char* param1, unsigned char* param2) +{ + static int index = 0; + static arcam_av_cc_t received_command; + static unsigned char received_param1; + static unsigned char received_param2; + + do { + static char buffer[8]; + ssize_t bytes = read(fd, buffer, sizeof buffer - index); + + if (bytes <= 0) + return -errno; + + char* cursor = buffer; + + while(bytes > 0) { + switch(index++) { + case 0: + if (*cursor != 'A') + index = 0; + break; + + case 1: + if (*cursor != 'V') { + index = 0; + continue; + } + break; + + case 2: + if (*cursor != '_') { + index = 0; + continue; + } + break; + + case 3: + received_command = *cursor; + break; + + case 4: + if (*cursor != ARCAM_AV_OK) { + index = 0; + continue; + } + break; + + case 5: + received_param1 = *cursor; + break; + + case 6: + received_param2 = *cursor; + break; + + case 7: + if (*cursor != 0x0D) { + index = 0; + continue; + } + break; + } + + ++cursor; + --bytes; + } + } while (index < 8); + + index = 0; + *command = received_command; + *param1 = received_param1; + *param2 = received_param2; + + return 0; +} + + +static int arcam_av_update(arcam_av_state_t* state, int fd) +{ + int result = -1; + + arcam_av_cc_t command = 0; + unsigned char param1 = 0; + unsigned char param2 = 0; + + while (!arcam_av_receive(fd, &command, &param1, &param2)) { + switch(command) { + case ARCAM_AV_POWER: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.power = param2; + result = 0; + break; + + case ARCAM_AV_ZONE2: + state->zone2.power = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_VOLUME_CHANGE: + case ARCAM_AV_VOLUME_SET: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.volume = param2; + result = 0; + break; + + case ARCAM_AV_ZONE2: + state->zone2.volume = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_MUTE: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.mute = param2; + result = 0; + break; + + case ARCAM_AV_ZONE2: + state->zone2.mute = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_DIRECT: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.direct = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_SOURCE: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.source = param2; + result = 0; + break; + + case ARCAM_AV_ZONE2: + state->zone2.source = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_SOURCE_TYPE: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.source_type = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_STEREO_DECODE: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.stereo_decode = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_STEREO_EFFECT: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.stereo_effect = param2; + result = 0; + break; + + default: + break; + } + break; + + case ARCAM_AV_MULTI_DECODE: + switch(param1) { + case ARCAM_AV_ZONE1: + state->zone1.multi_decode = param2; + result = 0; + break; + + default: + break; + } + break; + + default: + break; + } + } + + return result; +} + + +static void arcam_av_state_query(int fd) +{ + arcam_av_send(fd, ARCAM_AV_POWER, ARCAM_AV_ZONE1, ARCAM_AV_POWER_REQUEST); + arcam_av_send(fd, ARCAM_AV_VOLUME_CHANGE, ARCAM_AV_ZONE1, ARCAM_AV_VOLUME_REQUEST); + arcam_av_send(fd, ARCAM_AV_MUTE, ARCAM_AV_ZONE1, ARCAM_AV_MUTE_REQUEST); + arcam_av_send(fd, ARCAM_AV_DIRECT, ARCAM_AV_ZONE1, ARCAM_AV_DIRECT_REQUEST); + arcam_av_send(fd, ARCAM_AV_SOURCE, ARCAM_AV_ZONE1, ARCAM_AV_SOURCE_REQUEST); + arcam_av_send(fd, ARCAM_AV_SOURCE_TYPE, ARCAM_AV_ZONE1, ARCAM_AV_SOURCE_TYPE_REQUEST); + arcam_av_send(fd, ARCAM_AV_STEREO_DECODE, ARCAM_AV_ZONE1, ARCAM_AV_STEREO_DECODE_REQUEST); + arcam_av_send(fd, ARCAM_AV_MULTI_DECODE, ARCAM_AV_ZONE1, ARCAM_AV_MULTI_DECODE_REQUEST); + arcam_av_send(fd, ARCAM_AV_STEREO_EFFECT, ARCAM_AV_ZONE1, ARCAM_AV_STEREO_EFFECT_REQUEST); + + arcam_av_send(fd, ARCAM_AV_POWER, ARCAM_AV_ZONE2, ARCAM_AV_POWER_REQUEST); + arcam_av_send(fd, ARCAM_AV_VOLUME_CHANGE, ARCAM_AV_ZONE2, ARCAM_AV_VOLUME_REQUEST); + arcam_av_send(fd, ARCAM_AV_MUTE, ARCAM_AV_ZONE2, ARCAM_AV_MUTE_REQUEST); + arcam_av_send(fd, ARCAM_AV_SOURCE, ARCAM_AV_ZONE2, ARCAM_AV_SOURCE_REQUEST); +} + + +static int arcam_av_state_shm_id = -1; + +arcam_av_state_t* arcam_av_state_attach(const char* port) +{ + if (arcam_av_state_shm_id < 0) { + struct stat port_stat; + if (stat(port, &port_stat)) + return NULL; + + key_t ipc_key = ftok(port, 'A'); + if (ipc_key < 0) + return NULL; + + int shmflg = IPC_CREAT | (port_stat.st_mode & (S_IRWXU | S_IRWXG | S_IRWXO)); + arcam_av_state_shm_id = shmget(ipc_key, sizeof(arcam_av_state_t), shmflg); + if (arcam_av_state_shm_id < 0) + return NULL; + + struct shmid_ds shm_stat; + if (shmctl(arcam_av_state_shm_id, IPC_STAT, &shm_stat)) + return NULL; + + shm_stat.shm_perm.uid = port_stat.st_uid; + shm_stat.shm_perm.gid = port_stat.st_gid; + shmctl(arcam_av_state_shm_id, IPC_SET, &shm_stat); + } + + arcam_av_state_t* state = shmat(arcam_av_state_shm_id, NULL, 0); + + return (state == (void*)-1) ? NULL : state; +} + + +int arcam_av_state_detach(arcam_av_state_t* state) +{ + if (shmdt(state)) + return -1; + + struct shmid_ds shm_stat; + if (shmctl(arcam_av_state_shm_id, IPC_STAT, &shm_stat)) + return -1; + + if (!shm_stat.shm_nattch) { + shmctl(arcam_av_state_shm_id, IPC_RMID, NULL); + arcam_av_state_shm_id = -1; + } + + return 0; +} + + +static void arcam_av_server_broadcast(fd_set* fds, int fd_max, void* buffer, int bytes) +{ + int fd; + for (fd = 0; fd <= fd_max; ++fd) { + if (FD_ISSET(fd, fds)) { + send(fd, buffer, bytes, 0); + } + } +} + + +static int arcam_av_server_master(int server_fd) +{ + int result = 0; + + struct sockaddr_un server_address; + socklen_t server_address_length = sizeof(server_address) - 1; + if (getsockname(server_fd, (struct sockaddr*) &server_address, &server_address_length)) + return -errno; + + server_address.sun_path[server_address_length - offsetof(struct sockaddr_un, sun_path)] = '\0'; + + const char* port = server_address.sun_path + 1; + + int arcam_fd = arcam_av_connect(port); + + arcam_av_state_t* state = arcam_av_state_attach(port); + if (!state) { + close(arcam_fd); + return -errno; + } + + arcam_av_state_query(arcam_fd); + + fcntl(arcam_fd, F_SETFL, O_NONBLOCK); + + fd_set all_fds, client_fds, read_fds; + + FD_ZERO(&all_fds); + FD_ZERO(&client_fds); + FD_SET(arcam_fd, &all_fds); + FD_SET(server_fd, &all_fds); + int fd, fd_max = MAX(arcam_fd, server_fd); + + for(;;) { + read_fds = all_fds; + + if (select(fd_max + 1, &read_fds, NULL, NULL, NULL) < 0) { + perror("arcam_av_server_master(): select"); + result = -errno; + break; + } + + for(fd = fd_max; fd; --fd) { + if (FD_ISSET(fd, &read_fds)) { + if (fd == arcam_fd) { + if (arcam_av_update(state, arcam_fd)) + continue; + + arcam_av_server_broadcast(&client_fds, fd_max, "", 1); + } else if (fd == server_fd) { + struct sockaddr_un client_address; + socklen_t client_address_length = sizeof(client_address); + int client_fd = accept(server_fd, (struct sockaddr*) &client_address, &client_address_length); + + if (client_fd >= 0) { + FD_SET(client_fd, &all_fds); + FD_SET(client_fd, &client_fds); + fd_max = MAX(fd_max, client_fd); + } else { + perror("arcam_av_server_master(): accept"); + result = -errno; + goto exit; + } + } else { + pthread_t thread; + int bytes = recv(fd, &thread, sizeof(pthread_t), 0); + + if (bytes > 0) { + if (bytes == sizeof(pthread_t)) { + if (thread == pthread_self()) + goto exit; + + arcam_av_server_broadcast(&client_fds, fd_max, &thread, sizeof(pthread_t)); + } + } else { + close(fd); + FD_CLR(fd, &all_fds); + FD_CLR(fd, &client_fds); + fd_max -= (fd_max == fd); + } + } + } + } + } + +exit: + + for (fd = 0; fd <= fd_max; ++fd) { + if (fd != server_fd && FD_ISSET(fd, &all_fds)) { + close(fd); + } + } + + if (state) + arcam_av_state_detach(state); + + return result; +} + + +static int arcam_av_server_slave(int server_fd) +{ + for (;;) { + pthread_t thread; + int bytes = recv(server_fd, &thread, sizeof(pthread_t), 0); + + if (bytes > 0) { + if (bytes == sizeof(pthread_t)) + if (thread == pthread_self()) + return 0; + } else { + break; + } + } + + return -1; +} + + +static void* arcam_av_server_thread(void* context) +{ + sem_t* semaphore = context; + const char* port = *(const char**)(semaphore + 1); + + struct sockaddr_un address; + address.sun_family = AF_FILE; + address.sun_path[0] = '\0'; + strncpy(&address.sun_path[1], port, sizeof(address.sun_path) - 1); + int size = offsetof(struct sockaddr_un, sun_path) + + MIN(strlen(port) + 1, sizeof(address.sun_path)); + + signal(SIGPIPE, SIG_IGN); + + int quit = 0; + + while (!quit) { + int server_fd = socket(PF_UNIX, SOCK_STREAM, 0); + if (server_fd < 0) { + perror("arcam_av_server_thread(): socket"); + break; + } + + if (!bind(server_fd, (struct sockaddr*) &address, size)) { + if (!listen(server_fd, 10)) { + if (semaphore) { + sem_post(semaphore); + semaphore = NULL; + } + arcam_av_server_master(server_fd); + } else { + perror("arcam_av_server_master(): listen"); + } + quit = 1; + } else if (errno != EADDRINUSE) { + perror("arcam_av_server_thread(): bind"); + quit = 1; + } else if (!connect(server_fd, (struct sockaddr*) &address, size)) { + if (semaphore) { + sem_post(semaphore); + semaphore = NULL; + } + quit = !arcam_av_server_slave(server_fd); + } else { + perror("arcam_av_server_thread(): connect"); + quit = 1; + } + + close(server_fd); + } + + if (semaphore) + sem_post(semaphore); + + return NULL; +} + + +int arcam_av_server_start(pthread_t* thread, const char* port) +{ + struct { + sem_t semaphore; + const char* port; + } context; + + int result = 0; + + if (sem_init(&context.semaphore, 0, 0)) + return -1; + + context.port = port; + + if (pthread_create(thread, NULL, arcam_av_server_thread, &context)) + result = -1; + else + sem_wait(&context.semaphore); + + sem_destroy(&context.semaphore); + + return result; +} + + +int arcam_av_server_stop(pthread_t thread, const char* port) +{ + int client_fd = arcam_av_client(port); + if (client_fd < 0) + return -1; + + if (send(client_fd, &thread, sizeof(pthread_t), 0) > 0) + pthread_join(thread, NULL); + + close(client_fd); + return 0; +} + + +int arcam_av_client(const char* port) +{ + int client_fd = socket(PF_UNIX, SOCK_STREAM, 0); + if (client_fd < 0) + return -1; + + struct sockaddr_un address; + address.sun_family = AF_FILE; + address.sun_path[0] = '\0'; + strncpy(&address.sun_path[1], port, sizeof(address.sun_path) - 1); + int size = offsetof(struct sockaddr_un, sun_path) + + MIN(strlen(port) + 1, sizeof(address.sun_path)); + + const int max_retries = 5; + int retries = max_retries; + + do { + if (!connect(client_fd, (struct sockaddr*) &address, size)) + return client_fd; + + if (!retries--) + break; + + struct timeval sleep = {0, 10 * (max_retries - retries)}; + + select(0, NULL, NULL, NULL, &sleep); + + } while (errno == ECONNREFUSED); + + perror("arcam_av_client(): connect"); + + close(client_fd); + return -1; +} + diff -Nrup alsa-plugins-1.0.18-orig/arcam-av/arcam_av.h alsa-plugins-1.0.18/arcam-av/arcam_av.h --- alsa-plugins-1.0.18-orig/arcam-av/arcam_av.h 1970-01-01 01:00:00.000000000 +0100 +++ alsa-plugins-1.0.18/arcam-av/arcam_av.h 2009-01-02 22:30:58.000000000 +0000 @@ -0,0 +1,161 @@ +/* + * ALSA -> Arcam AV control plugin + * + * Copyright (c) 2009 by Peter Stokes <linux(a)dadeos.co.uk> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <pthread.h> + +typedef enum { + ARCAM_AV_ZONE1 = '1', + ARCAM_AV_ZONE2 = '2' +} arcam_av_zone_t; + + +typedef enum { + ARCAM_AV_POWER = '*', + ARCAM_AV_VOLUME_CHANGE = '/', + ARCAM_AV_VOLUME_SET = '0', + ARCAM_AV_MUTE = '.', + ARCAM_AV_SOURCE = '1', + ARCAM_AV_SOURCE_TYPE = '7', + ARCAM_AV_DIRECT = '3', + ARCAM_AV_STEREO_DECODE = '4', + ARCAM_AV_MULTI_DECODE = '5', + ARCAM_AV_STEREO_EFFECT = '6' +} arcam_av_cc_t; + + +typedef enum { + ARCAM_AV_OK = 'P', + ARCAM_AV_ERROR = 'R' +} arcam_av_rc_t; + + +typedef enum { + ARCAM_AV_POWER_STAND_BY = '0', + ARCAM_AV_POWER_ON = '1', + ARCAM_AV_POWER_REQUEST = '9' +} arcam_av_power_t; + + +typedef enum { + ARCAM_AV_VOLUME_MIN = '0', + ARCAM_AV_VOLUME_REQUEST = '9' +} arcam_av_volume_t; + + +typedef enum { + ARCAM_AV_MUTE_ON = '0', + ARCAM_AV_MUTE_OFF = '1', + ARCAM_AV_MUTE_REQUEST = '9' +} arcam_av_mute_t; + + +typedef enum { + ARCAM_AV_DIRECT_DISABLE = '0', + ARCAM_AV_DIRECT_ENABLE = '1', + ARCAM_AV_DIRECT_REQUEST = '9' +} arcam_av_direct_t; + + +typedef enum { + ARCAM_AV_SOURCE_DVD = '0', + ARCAM_AV_SOURCE_SAT = '1', + ARCAM_AV_SOURCE_AV = '2', + ARCAM_AV_SOURCE_PVR = '3', + ARCAM_AV_SOURCE_VCR = '4', + ARCAM_AV_SOURCE_CD = '5', + ARCAM_AV_SOURCE_FM = '6', + ARCAM_AV_SOURCE_AM = '7', + ARCAM_AV_SOURCE_DVDA = '8', + ARCAM_AV_SOURCE_REQUEST = '9' +} arcam_av_source_t; + + +typedef enum { + ARCAM_AV_SOURCE_TYPE_ANALOGUE = '0', + ARCAM_AV_SOURCE_TYPE_DIGITAL = '1', + ARCAM_AV_SOURCE_TYPE_REQUEST = '9' +} arcam_av_source_type_t; + + +typedef enum { + ARCAM_AV_STEREO_DECODE_MONO = '.', + ARCAM_AV_STEREO_DECODE_STEREO = '/', + ARCAM_AV_STEREO_DECODE_PLII_MOVIE = '0', + ARCAM_AV_STEREO_DECODE_PLII_MUSIC = '1', + ARCAM_AV_STEREO_DECODE_PLIIx_MOVIE = '3', + ARCAM_AV_STEREO_DECODE_PLIIx_MUSIC = '4', + ARCAM_AV_STEREO_DECODE_DOLBY_PL = '6', + ARCAM_AV_STEREO_DECODE_NEO6_CINEMA = '7', + ARCAM_AV_STEREO_DECODE_NEO6_MUSIC = '8', + ARCAM_AV_STEREO_DECODE_REQUEST = '9' +} arcam_av_stereo_decode_t; + + +typedef enum { + ARCAM_AV_MULTI_DECODE_MONO = '.', + ARCAM_AV_MULTI_DECODE_STEREO = '/', + ARCAM_AV_MULTI_DECODE_MULTI_CHANNEL = '0', + ARCAM_AV_MULTI_DECODE_PLIIx = '2', + ARCAM_AV_MULTI_DECODE_REQUEST = '9' +} arcam_av_multi_decode_t; + + +typedef enum { + ARCAM_AV_STEREO_EFFECT_NONE = '0', + ARCAM_AV_STEREO_EFFECT_MUSIC = '1', + ARCAM_AV_STEREO_EFFECT_PARTY = '2', + ARCAM_AV_STEREO_EFFECT_CLUB = '3', + ARCAM_AV_STEREO_EFFECT_HALL = '4', + ARCAM_AV_STEREO_EFFECT_SPORTS = '5', + ARCAM_AV_STEREO_EFFECT_CHURCH = '6', + ARCAM_AV_STEREO_EFFECT_REQUEST = '9' +} arcam_av_stereo_effect_t; + +int arcam_av_connect(const char* port); +int arcam_av_send(int fd, arcam_av_cc_t command, unsigned char param1, unsigned char param2); + + +typedef struct arcam_av_state { + struct { + arcam_av_power_t power; + unsigned char volume; + arcam_av_mute_t mute; + arcam_av_direct_t direct; + arcam_av_source_t source; + arcam_av_source_type_t source_type; + arcam_av_stereo_decode_t stereo_decode; + arcam_av_stereo_effect_t stereo_effect; + arcam_av_multi_decode_t multi_decode; + } zone1; + struct { + arcam_av_power_t power; + unsigned char volume; + arcam_av_mute_t mute; + arcam_av_source_t source; + } zone2; +} arcam_av_state_t; + +arcam_av_state_t* arcam_av_state_attach(const char* port); +int arcam_av_state_detach(arcam_av_state_t* state); + +int arcam_av_server_start(pthread_t* thread, const char* port); +int arcam_av_server_stop(pthread_t thread, const char* port); + +int arcam_av_client(const char* port); diff -Nrup alsa-plugins-1.0.18-orig/arcam-av/ctl_arcam_av.c alsa-plugins-1.0.18/arcam-av/ctl_arcam_av.c --- alsa-plugins-1.0.18-orig/arcam-av/ctl_arcam_av.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-plugins-1.0.18/arcam-av/ctl_arcam_av.c 2009-01-02 22:30:55.000000000 +0000 @@ -0,0 +1,1024 @@ +/* + * ALSA -> Arcam AV control plugin + * + * Copyright (c) 2009 by Peter Stokes <linux(a)dadeos.co.uk> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sys/socket.h> +#include <alsa/asoundlib.h> +#include <alsa/control_external.h> + +#include "arcam_av.h" + + +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) +#define MIN(a, b) ((a) < (b) ? (a) : (b)) +#define MID(a, b, c) ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \ + : ((b) > (c) ? (b) : ((a) > (c) ? (c) : (a)))) + + +static const char* arcam_av_name = "Arcam AV"; +static const char* arcam_av_power_name = "Power Switch"; +static const char* arcam_av_volume_name = "Master Playback Volume"; +static const char* arcam_av_mute_name = "Master Playback Switch"; +static const char* arcam_av_direct_name = "Direct Playback Switch"; +static const char* arcam_av_source_name = "Source Playback Route"; +static const char* arcam_av_source_type_name = "Source Type Playback Route"; +static const char* arcam_av_stereo_decode_name = "Stereo Decode Playback Route"; +static const char* arcam_av_multi_decode_name = "Multi-Channel Decode Playback Route"; +static const char* arcam_av_stereo_effect_name = "Stereo Effect Playback Route"; + +static const struct { + arcam_av_source_t code; + const char* name; +} arcam_av_sources[] = { + {ARCAM_AV_SOURCE_DVD, "DVD" }, + {ARCAM_AV_SOURCE_SAT, "SAT" }, + {ARCAM_AV_SOURCE_AV, "AV" }, + {ARCAM_AV_SOURCE_PVR, "PVR" }, + {ARCAM_AV_SOURCE_VCR, "VCR" }, + {ARCAM_AV_SOURCE_CD, "CD" }, + {ARCAM_AV_SOURCE_FM, "FM" }, + {ARCAM_AV_SOURCE_AM, "AM" }, + {ARCAM_AV_SOURCE_DVDA, "DVDA" } +}; + +static const struct { + arcam_av_source_type_t code; + const char* name; +} arcam_av_source_types[] = { + {ARCAM_AV_SOURCE_TYPE_ANALOGUE, "Analogue" }, + {ARCAM_AV_SOURCE_TYPE_DIGITAL, "Digital" } +}; + +static const struct { + arcam_av_direct_t code; + const char* name; +} arcam_av_direct[] = { + {ARCAM_AV_DIRECT_DISABLE, "Disable" }, + {ARCAM_AV_DIRECT_ENABLE, "Enable" } +}; + +static const struct { + arcam_av_stereo_decode_t code; + const char* name; +} arcam_av_stereo_decode_modes[] = { + {ARCAM_AV_STEREO_DECODE_MONO, "Mono" }, + {ARCAM_AV_STEREO_DECODE_STEREO, "Stereo" }, + {ARCAM_AV_STEREO_DECODE_PLII_MOVIE, "Pro Logic II Movie" }, + {ARCAM_AV_STEREO_DECODE_PLII_MUSIC, "Pro Logic II Music" }, + {ARCAM_AV_STEREO_DECODE_PLIIx_MOVIE, "Pro Logic IIx Movie" }, + {ARCAM_AV_STEREO_DECODE_PLIIx_MUSIC, "Pro Logic IIx Music" }, + {ARCAM_AV_STEREO_DECODE_DOLBY_PL, "Dolby Pro Logic" }, + {ARCAM_AV_STEREO_DECODE_NEO6_CINEMA, "Neo:6 Cinema" }, + {ARCAM_AV_STEREO_DECODE_NEO6_MUSIC, "Neo:6 Music" } +}; + +static const struct { + arcam_av_multi_decode_t code; + const char* name; +} arcam_av_multi_decode_modes[] = { + {ARCAM_AV_MULTI_DECODE_MONO, "Mono down-mix" }, + {ARCAM_AV_MULTI_DECODE_STEREO, "Stereo down-mix" }, + {ARCAM_AV_MULTI_DECODE_MULTI_CHANNEL, "Multi-channel" }, + {ARCAM_AV_MULTI_DECODE_PLIIx, "Pro Logic IIx" } +}; + +static const struct { + arcam_av_stereo_effect_t code; + const char* name; +} arcam_av_stereo_effects[] = { + {ARCAM_AV_STEREO_EFFECT_NONE, "None" }, + {ARCAM_AV_STEREO_EFFECT_MUSIC, "Music" }, + {ARCAM_AV_STEREO_EFFECT_PARTY, "Party" }, + {ARCAM_AV_STEREO_EFFECT_CLUB, "Club" }, + {ARCAM_AV_STEREO_EFFECT_HALL, "Hall" }, + {ARCAM_AV_STEREO_EFFECT_SPORTS, "Sports" }, + {ARCAM_AV_STEREO_EFFECT_CHURCH, "Church" } +}; + + +typedef struct snd_ctl_arcam_av { + snd_ctl_ext_t ext; + int port_fd; + int shm_id; + const char* port; + arcam_av_zone_t zone; + arcam_av_state_t local; + arcam_av_state_t* global; + pthread_t server; +} snd_ctl_arcam_av_t; + +static void arcam_av_close(snd_ctl_ext_t *ext) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + if (arcam_av->port_fd >= 0) + close(arcam_av->port_fd); + + if (arcam_av->global) + arcam_av_state_detach(arcam_av->global); + + if (arcam_av->ext.poll_fd >= 0) { + close(arcam_av->ext.poll_fd); + arcam_av_server_stop(arcam_av->server, arcam_av->port); + } + + free(arcam_av); +} + +static int arcam_av_elem_count(snd_ctl_ext_t *ext) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + return arcam_av->zone == ARCAM_AV_ZONE1 ? 9 : 4; +} + +static int arcam_av_elem_list(snd_ctl_ext_t *ext, unsigned int offset, snd_ctl_elem_id_t *id) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); + + if (arcam_av->zone == ARCAM_AV_ZONE1) { + switch(offset) { + case 0: + snd_ctl_elem_id_set_name(id, arcam_av_power_name); + break; + + case 1: + snd_ctl_elem_id_set_name(id, arcam_av_volume_name); + break; + + case 2: + snd_ctl_elem_id_set_name(id, arcam_av_mute_name); + break; + + case 3: + snd_ctl_elem_id_set_name(id, arcam_av_direct_name); + break; + + case 4: + snd_ctl_elem_id_set_name(id, arcam_av_source_name); + break; + + case 5: + snd_ctl_elem_id_set_name(id, arcam_av_source_type_name); + break; + + case 6: + snd_ctl_elem_id_set_name(id, arcam_av_stereo_decode_name); + break; + + case 7: + snd_ctl_elem_id_set_name(id, arcam_av_multi_decode_name); + break; + + case 8: + snd_ctl_elem_id_set_name(id, arcam_av_stereo_effect_name); + break; + } + } else { + switch(offset) { + case 0: + snd_ctl_elem_id_set_name(id, arcam_av_power_name); + break; + + case 1: + snd_ctl_elem_id_set_name(id, arcam_av_volume_name); + break; + + case 2: + snd_ctl_elem_id_set_name(id, arcam_av_mute_name); + break; + + case 3: + snd_ctl_elem_id_set_name(id, arcam_av_source_name); + break; + } + } + + return 0; +} + +static snd_ctl_ext_key_t arcam_av_find_elem(snd_ctl_ext_t *ext ATTRIBUTE_UNUSED, + const snd_ctl_elem_id_t *id) +{ + const char *name; + + name = snd_ctl_elem_id_get_name(id); + if (!strcmp(name, arcam_av_power_name)) { + return ARCAM_AV_POWER; + } else if (!strcmp(name, arcam_av_volume_name)) { + return ARCAM_AV_VOLUME_SET; + } else if (!strcmp(name, arcam_av_mute_name)) { + return ARCAM_AV_MUTE; + } else if (!strcmp(name, arcam_av_direct_name)) { + return ARCAM_AV_DIRECT; + } else if (!strcmp(name, arcam_av_source_name)) { + return ARCAM_AV_SOURCE; + } else if (!strcmp(name, arcam_av_source_type_name)) { + return ARCAM_AV_SOURCE_TYPE; + } else if (!strcmp(name, arcam_av_stereo_decode_name)) { + return ARCAM_AV_STEREO_DECODE; + } else if (!strcmp(name, arcam_av_multi_decode_name)) { + return ARCAM_AV_MULTI_DECODE; + } else if (!strcmp(name, arcam_av_stereo_effect_name)) { + return ARCAM_AV_STEREO_EFFECT; + } + + return SND_CTL_EXT_KEY_NOT_FOUND; +} + +static int arcam_av_get_attribute(snd_ctl_ext_t *ext ATTRIBUTE_UNUSED, + snd_ctl_ext_key_t key, int *type, + unsigned int *acc, unsigned int *count) +{ + switch(key) { + case ARCAM_AV_POWER: + *type = SND_CTL_ELEM_TYPE_BOOLEAN; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_VOLUME_SET: + *type = SND_CTL_ELEM_TYPE_INTEGER; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_MUTE: + *type = SND_CTL_ELEM_TYPE_BOOLEAN; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_DIRECT: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_SOURCE: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_SOURCE_TYPE: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_STEREO_DECODE: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_MULTI_DECODE: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + case ARCAM_AV_STEREO_EFFECT: + *type = SND_CTL_ELEM_TYPE_ENUMERATED; + *acc = SND_CTL_EXT_ACCESS_READWRITE; + *count = 1; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int arcam_av_get_integer_info(snd_ctl_ext_t *ext, + snd_ctl_ext_key_t key, + long *imin, long *imax, long *istep) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + switch(key) { + case ARCAM_AV_VOLUME_SET: + *istep = 1; + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + *imin = 0; + *imax = 100; + break; + + case ARCAM_AV_ZONE2: + *imin = 20; + *imax = 83; + break; + } + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int arcam_av_get_enumerated_info(snd_ctl_ext_t *ext ATTRIBUTE_UNUSED, + snd_ctl_ext_key_t key, + unsigned int *items) +{ + switch(key) { + case ARCAM_AV_SOURCE: + *items = ARRAY_SIZE(arcam_av_sources); + break; + + case ARCAM_AV_SOURCE_TYPE: + *items = ARRAY_SIZE(arcam_av_source_types); + break; + + case ARCAM_AV_DIRECT: + *items = ARRAY_SIZE(arcam_av_direct); + break; + + case ARCAM_AV_STEREO_DECODE: + *items = ARRAY_SIZE(arcam_av_stereo_decode_modes); + break; + + case ARCAM_AV_MULTI_DECODE: + *items = ARRAY_SIZE(arcam_av_multi_decode_modes); + break; + + case ARCAM_AV_STEREO_EFFECT: + *items = ARRAY_SIZE(arcam_av_stereo_effects); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int arcam_av_get_enumerated_name(snd_ctl_ext_t *ext ATTRIBUTE_UNUSED, + snd_ctl_ext_key_t key, + unsigned int item, char *name, + size_t name_max_len) +{ + const char* label; + + switch(key) { + case ARCAM_AV_SOURCE: + if (item >= ARRAY_SIZE(arcam_av_sources)) + return -EINVAL; + + label = arcam_av_sources[item].name; + break; + + case ARCAM_AV_SOURCE_TYPE: + if (item >= ARRAY_SIZE(arcam_av_source_types)) + return -EINVAL; + + label = arcam_av_source_types[item].name; + break; + + case ARCAM_AV_DIRECT: + if (item >= ARRAY_SIZE(arcam_av_direct)) + return -EINVAL; + + label = arcam_av_direct[item].name; + break; + + case ARCAM_AV_STEREO_DECODE: + if (item >= ARRAY_SIZE(arcam_av_stereo_decode_modes)) + return -EINVAL; + + label = arcam_av_stereo_decode_modes[item].name; + break; + + case ARCAM_AV_MULTI_DECODE: + if (item >= ARRAY_SIZE(arcam_av_multi_decode_modes)) + return -EINVAL; + + label = arcam_av_multi_decode_modes[item].name; + break; + + case ARCAM_AV_STEREO_EFFECT: + if (item >= ARRAY_SIZE(arcam_av_stereo_effects)) + return -EINVAL; + + label = arcam_av_stereo_effects[item].name; + break; + + default: + return -EINVAL; + } + + strncpy(name, label, name_max_len - 1); + name[name_max_len - 1] = '\0'; + + return 0; +} + +static int arcam_av_read_integer(snd_ctl_ext_t *ext, snd_ctl_ext_key_t key, long *value) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + switch(key) { + case ARCAM_AV_POWER: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.power = arcam_av->global->zone1.power; + *value = !(arcam_av->local.zone1.power == ARCAM_AV_POWER_STAND_BY); + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.power = arcam_av->global->zone2.power; + *value = !(arcam_av->local.zone2.power == ARCAM_AV_POWER_STAND_BY); + break; + } + break; + + case ARCAM_AV_VOLUME_SET: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.volume = arcam_av->global->zone1.volume; + *value = MID(0, arcam_av->local.zone1.volume - ARCAM_AV_VOLUME_MIN, 100); + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.volume = arcam_av->global->zone2.volume; + *value = MID(20, arcam_av->local.zone2.volume - ARCAM_AV_VOLUME_MIN, 83); + break; + } + break; + + case ARCAM_AV_MUTE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.mute = arcam_av->global->zone1.mute; + *value = !(arcam_av->local.zone1.mute == ARCAM_AV_MUTE_ON); + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.mute = arcam_av->global->zone2.mute; + *value = !(arcam_av->local.zone2.mute == ARCAM_AV_MUTE_ON); + break; + } + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int arcam_av_read_enumerated(snd_ctl_ext_t *ext, + snd_ctl_ext_key_t key, + unsigned int *item) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + unsigned int i; + + switch(key) { + case ARCAM_AV_SOURCE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.source = arcam_av->global->zone1.source; + for (i = 0; i < ARRAY_SIZE(arcam_av_sources); ++i) { + if (arcam_av_sources[i].code == arcam_av->local.zone1.source) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.source = arcam_av->global->zone2.source; + for (i = 0; i < ARRAY_SIZE(arcam_av_sources); ++i) { + if (arcam_av_sources[i].code == arcam_av->local.zone2.source) { + *item = i; + break; + } + } + break; + } + break; + + case ARCAM_AV_SOURCE_TYPE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.source_type = arcam_av->global->zone1.source_type; + for (i = 0; i < ARRAY_SIZE(arcam_av_source_types); ++i) { + if (arcam_av_source_types[i].code == arcam_av->local.zone1.source_type) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_DIRECT: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.direct = arcam_av->global->zone1.direct; + for (i = 0; i < ARRAY_SIZE(arcam_av_direct); ++i) { + if (arcam_av_direct[i].code == arcam_av->local.zone1.direct) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_STEREO_DECODE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.stereo_decode = arcam_av->global->zone1.stereo_decode; + for (i = 0; i < ARRAY_SIZE(arcam_av_stereo_decode_modes); ++i) { + if (arcam_av_stereo_decode_modes[i].code == arcam_av->local.zone1.stereo_decode) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_STEREO_EFFECT: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.stereo_effect = arcam_av->global->zone1.stereo_effect; + for (i = 0; i < ARRAY_SIZE(arcam_av_stereo_effects); ++i) { + if (arcam_av_stereo_effects[i].code == arcam_av->local.zone1.stereo_effect) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_MULTI_DECODE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.multi_decode = arcam_av->global->zone1.multi_decode; + for (i = 0; i < ARRAY_SIZE(arcam_av_multi_decode_modes); ++i) { + if (arcam_av_multi_decode_modes[i].code == arcam_av->local.zone1.multi_decode) { + *item = i; + break; + } + } + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int arcam_av_write_integer(snd_ctl_ext_t *ext, snd_ctl_ext_key_t key, long *value) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + unsigned char volume = ARCAM_AV_VOLUME_MIN; + + switch(key) { + case ARCAM_AV_POWER: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.power = ARCAM_AV_POWER_STAND_BY + *value; + if (arcam_av->global->zone1.power == arcam_av->local.zone1.power) + return 0; + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.power = ARCAM_AV_POWER_STAND_BY + *value; + if (arcam_av->global->zone2.power == arcam_av->local.zone2.power) + return 0; + break; + } + break; + + case ARCAM_AV_VOLUME_SET: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.volume = ARCAM_AV_VOLUME_MIN + *value; + if (arcam_av->global->zone1.volume == arcam_av->local.zone1.volume) + return 0; + + if (arcam_av->global->zone1.mute == ARCAM_AV_MUTE_ON) { + arcam_av->global->zone1.volume = arcam_av->local.zone1.volume; + return 1; + } + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.volume = ARCAM_AV_VOLUME_MIN + *value; + if (arcam_av->global->zone2.volume == arcam_av->local.zone2.volume) + return 0; + + if (arcam_av->global->zone2.mute == ARCAM_AV_MUTE_ON) { + arcam_av->global->zone2.volume = arcam_av->local.zone2.volume; + return 1; + } + break; + } + break; + + case ARCAM_AV_MUTE: + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.mute = ARCAM_AV_MUTE_ON + *value; + if (arcam_av->global->zone1.mute == arcam_av->local.zone1.mute) + return 0; + + volume = arcam_av->global->zone1.volume; + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.mute = ARCAM_AV_MUTE_ON + *value; + if (arcam_av->global->zone2.mute == arcam_av->local.zone2.mute) + return 0; + + volume = arcam_av->global->zone2.volume; + break; + } + + if (*value) + arcam_av_send(arcam_av->port_fd, ARCAM_AV_VOLUME_SET, arcam_av->zone, volume); + break; + + default: + return -EINVAL; + } + + if (!arcam_av_send(arcam_av->port_fd, key, arcam_av->zone, '0' + *value)) + return 1; + else + return -1; +} + +static int arcam_av_write_enumerated(snd_ctl_ext_t *ext, snd_ctl_ext_key_t key, unsigned int *item) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + char code; + + switch(key) { + case ARCAM_AV_SOURCE: + if (*item >= ARRAY_SIZE(arcam_av_sources)) + return -EINVAL; + + code = arcam_av_sources[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.source = code; + if (arcam_av->global->zone1.source == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + arcam_av->local.zone2.source = code; + if (arcam_av->global->zone2.source == code) + return 0; + break; + } + break; + + case ARCAM_AV_SOURCE_TYPE: + if (*item >= ARRAY_SIZE(arcam_av_source_types)) + return -EINVAL; + + code = arcam_av_source_types[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.source_type = code; + if (arcam_av->global->zone1.source_type == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_DIRECT: + if (*item >= ARRAY_SIZE(arcam_av_direct)) + return -EINVAL; + + code = arcam_av_direct[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.direct = code; + if (arcam_av->global->zone1.direct == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_STEREO_DECODE: + if (*item >= ARRAY_SIZE(arcam_av_stereo_decode_modes)) + return -EINVAL; + + code = arcam_av_stereo_decode_modes[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.stereo_decode = code; + if (arcam_av->global->zone1.stereo_decode == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_STEREO_EFFECT: + if (*item >= ARRAY_SIZE(arcam_av_stereo_effects)) + return -EINVAL; + + code = arcam_av_stereo_effects[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.stereo_effect = code; + if (arcam_av->global->zone1.stereo_effect == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + case ARCAM_AV_MULTI_DECODE: + if (*item >= ARRAY_SIZE(arcam_av_multi_decode_modes)) + return -EINVAL; + + code = arcam_av_multi_decode_modes[*item].code; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + arcam_av->local.zone1.multi_decode = code; + if (arcam_av->global->zone1.multi_decode == code) + return 0; + break; + + case ARCAM_AV_ZONE2: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + if (!arcam_av_send(arcam_av->port_fd, key, arcam_av->zone, code)) + return 1; + else + return -1; +} + + +static int arcam_av_read_event(snd_ctl_ext_t *ext, snd_ctl_elem_id_t *id, unsigned int *event_mask) +{ + snd_ctl_arcam_av_t *arcam_av = ext->private_data; + + switch(arcam_av->zone) { + case ARCAM_AV_ZONE1: + if (arcam_av->local.zone1.power != arcam_av->global->zone1.power) { + snd_ctl_elem_id_set_name(id, arcam_av_power_name); + arcam_av->local.zone1.power = arcam_av->global->zone1.power; + } else if (arcam_av->local.zone1.volume != arcam_av->global->zone1.volume) { + snd_ctl_elem_id_set_name(id, arcam_av_volume_name); + arcam_av->local.zone1.volume = arcam_av->global->zone1.volume; + } else if (arcam_av->local.zone1.mute != arcam_av->global->zone1.mute) { + snd_ctl_elem_id_set_name(id, arcam_av_mute_name); + arcam_av->local.zone1.mute = arcam_av->global->zone1.mute; + } else if (arcam_av->local.zone1.direct != arcam_av->global->zone1.direct) { + snd_ctl_elem_id_set_name(id, arcam_av_direct_name); + arcam_av->local.zone1.direct = arcam_av->global->zone1.direct; + } else if (arcam_av->local.zone1.source != arcam_av->global->zone1.source) { + snd_ctl_elem_id_set_name(id, arcam_av_source_name); + arcam_av->local.zone1.source = arcam_av->global->zone1.source; + } else if (arcam_av->local.zone1.source_type != arcam_av->global->zone1.source_type) { + snd_ctl_elem_id_set_name(id, arcam_av_source_type_name); + arcam_av->local.zone1.source_type = arcam_av->global->zone1.source_type; + } else if (arcam_av->local.zone1.stereo_decode != arcam_av->global->zone1.stereo_decode) { + snd_ctl_elem_id_set_name(id, arcam_av_stereo_decode_name); + arcam_av->local.zone1.stereo_decode = arcam_av->global->zone1.stereo_decode; + } else if (arcam_av->local.zone1.stereo_effect != arcam_av->global->zone1.stereo_effect) { + snd_ctl_elem_id_set_name(id, arcam_av_stereo_effect_name); + arcam_av->local.zone1.stereo_effect = arcam_av->global->zone1.stereo_effect; + } else if (arcam_av->local.zone1.multi_decode != arcam_av->global->zone1.multi_decode) { + snd_ctl_elem_id_set_name(id, arcam_av_multi_decode_name); + arcam_av->local.zone1.multi_decode = arcam_av->global->zone1.multi_decode; + } else { + char buf[10]; + if (recv(arcam_av->ext.poll_fd, buf, sizeof(buf), 0) <= 0) { + close(arcam_av->ext.poll_fd); + arcam_av->ext.poll_fd = arcam_av_client(arcam_av->port); + if (arcam_av->ext.poll_fd > 0) + fcntl(arcam_av->ext.poll_fd, F_SETFL, O_NONBLOCK); + } + + return -EAGAIN; + } + break; + + case ARCAM_AV_ZONE2: + if (arcam_av->local.zone2.power != arcam_av->global->zone2.power) { + snd_ctl_elem_id_set_name(id, arcam_av_power_name); + arcam_av->local.zone2.power = arcam_av->global->zone2.power; + } else if (arcam_av->local.zone2.volume != arcam_av->global->zone2.volume) { + snd_ctl_elem_id_set_name(id, arcam_av_volume_name); + arcam_av->local.zone2.volume = arcam_av->global->zone2.volume; + } else if (arcam_av->local.zone2.mute != arcam_av->global->zone2.mute) { + snd_ctl_elem_id_set_name(id, arcam_av_mute_name); + arcam_av->local.zone2.mute = arcam_av->global->zone2.mute; + } else if (arcam_av->local.zone2.source != arcam_av->global->zone2.source) { + snd_ctl_elem_id_set_name(id, arcam_av_source_name); + arcam_av->local.zone2.source = arcam_av->global->zone2.source; + } else { + char buf[10]; + if (recv(arcam_av->ext.poll_fd, buf, sizeof(buf), 0) <= 0) { + close(arcam_av->ext.poll_fd); + arcam_av->ext.poll_fd = arcam_av_client(arcam_av->port); + if (arcam_av->ext.poll_fd > 0) + fcntl(arcam_av->ext.poll_fd, F_SETFL, O_NONBLOCK); + } + + return -EAGAIN; + } + break; + } + + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); + + *event_mask = SND_CTL_EVENT_MASK_VALUE; + + return 1; +} + + +static snd_ctl_ext_callback_t arcam_av_ext_callback = { + .close = arcam_av_close, + .elem_count = arcam_av_elem_count, + .elem_list = arcam_av_elem_list, + .find_elem = arcam_av_find_elem, + .get_attribute = arcam_av_get_attribute, + .get_integer_info = arcam_av_get_integer_info, + .get_enumerated_info = arcam_av_get_enumerated_info, + .get_enumerated_name = arcam_av_get_enumerated_name, + .read_integer = arcam_av_read_integer, + .read_enumerated = arcam_av_read_enumerated, + .write_integer = arcam_av_write_integer, + .write_enumerated = arcam_av_write_enumerated, + .read_event = arcam_av_read_event, +}; + + +SND_CTL_PLUGIN_DEFINE_FUNC(arcam_av) +{ + snd_config_iterator_t it, next; + const char *port = "/dev/ttyS0"; + long zone = 1; + int err; + snd_ctl_arcam_av_t *arcam_av = NULL; + + snd_config_for_each(it, next, conf) { + snd_config_t *n = snd_config_iterator_entry(it); + const char *id; + if (snd_config_get_id(n, &id) < 0) + continue; + if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0) + continue; + if (strcmp(id, "port") == 0) { + if (snd_config_get_string(n, &port) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "zone") == 0) { + if (snd_config_get_integer(n, &zone) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + if (zone < 1 || zone > 2) { + SNDERR("Invalid value for %s", id); + return -EINVAL; + } + continue; + } + SNDERR("Unknown field %s", id); + return -EINVAL; + } + + if (access(port, R_OK | W_OK) < 0) { + err = -errno; + goto error; + } + + arcam_av = calloc(1, sizeof(*arcam_av) + strlen(port) + 1); + + if (!arcam_av) + return -ENOMEM; + + arcam_av->ext.version = SND_CTL_EXT_VERSION; + arcam_av->ext.card_idx = 0; + strncpy(arcam_av->ext.id, arcam_av_name, sizeof(arcam_av->ext.id) - 1); + strncpy(arcam_av->ext.name, arcam_av_name, sizeof(arcam_av->ext.name) - 1); + strncpy(arcam_av->ext.longname, arcam_av_name, sizeof(arcam_av->ext.longname) - 1); + strncpy(arcam_av->ext.mixername, arcam_av_name, sizeof(arcam_av->ext.mixername) - 1); + arcam_av->ext.poll_fd = -1; + arcam_av->ext.callback = &arcam_av_ext_callback; + arcam_av->ext.private_data = arcam_av; + + arcam_av->shm_id= -1; + arcam_av->port_fd = -1; + arcam_av->port = strcpy((char*)(arcam_av + 1), port); + arcam_av->zone = zone != 2 ? ARCAM_AV_ZONE1 : ARCAM_AV_ZONE2; + + arcam_av->port_fd = arcam_av_connect(arcam_av->port); + if (arcam_av->port_fd < 0) { + err = -errno; + goto error; + } + + if (arcam_av_server_start(&arcam_av->server, arcam_av->port)) { + err = -errno; + goto error; + } + + arcam_av->ext.poll_fd = arcam_av_client(arcam_av->port); + if (arcam_av->ext.poll_fd < 0) { + err = -errno; + goto error; + } + + fcntl(arcam_av->ext.poll_fd, F_SETFL, O_NONBLOCK); + + arcam_av->global = arcam_av_state_attach(arcam_av->port); + if (!arcam_av->global) { + err = -errno; + goto error; + } + + err = snd_ctl_ext_create(&arcam_av->ext, name, mode); + if (err < 0) + goto error; + + *handlep = arcam_av->ext.handle; + return 0; + + error: + perror("arcam_av()"); + arcam_av_close(&arcam_av->ext); + return err; +} + +SND_CTL_PLUGIN_SYMBOL(arcam_av); diff -Nrup alsa-plugins-1.0.18-orig/configure.in alsa-plugins-1.0.18/configure.in --- alsa-plugins-1.0.18-orig/configure.in 2008-10-29 12:42:13.000000000 +0000 +++ alsa-plugins-1.0.18/configure.in 2009-01-02 22:25:12.000000000 +0000 @@ -140,6 +140,7 @@ AC_OUTPUT([ maemo/Makefile doc/Makefile usb_stream/Makefile + arcam-av/Makefile ]) dnl Show the build conditions diff -Nrup alsa-plugins-1.0.18-orig/doc/Makefile.am alsa-plugins-1.0.18/doc/Makefile.am --- alsa-plugins-1.0.18-orig/doc/Makefile.am 2008-10-29 12:42:13.000000000 +0000 +++ alsa-plugins-1.0.18/doc/Makefile.am 2009-01-02 23:19:19.000000000 +0000 @@ -1,3 +1,3 @@ EXTRA_DIST = README-pcm-oss README-jack README-pulse README-maemo \ upmix.txt vdownmix.txt samplerate.txt a52.txt lavcrate.txt \ - speexrate.txt + speexrate.txt README-arcam-av diff -Nrup alsa-plugins-1.0.18-orig/doc/Makefile.in alsa-plugins-1.0.18/doc/Makefile.in --- alsa-plugins-1.0.18-orig/doc/Makefile.in 2008-10-29 12:47:43.000000000 +0000 +++ alsa-plugins-1.0.18/doc/Makefile.in 2009-01-02 23:20:03.000000000 +0000 @@ -182,7 +182,7 @@ sysconfdir = @sysconfdir@ target_alias = @target_alias@ EXTRA_DIST = README-pcm-oss README-jack README-pulse README-maemo \ upmix.txt vdownmix.txt samplerate.txt a52.txt lavcrate.txt \ - speexrate.txt + speexrate.txt README-arcam-av all: all-am diff -Nrup alsa-plugins-1.0.18-orig/doc/README-arcam-av alsa-plugins-1.0.18/doc/README-arcam-av --- alsa-plugins-1.0.18-orig/doc/README-arcam-av 1970-01-01 01:00:00.000000000 +0100 +++ alsa-plugins-1.0.18/doc/README-arcam-av 2009-01-02 23:18:14.000000000 +0000 @@ -0,0 +1,29 @@ +Arcam AV Amplifier ALSA Control plugin +====================================== + +This plugin exposes the controls for an Arcam AV amplifier +(see: http://www.arcam.co.uk/) as an ALSA mixer device. + +To use this plugin you will need to connect the amplifier +to the PC using an RS-232 null-modem cable and use the +following ALSA configuration: + + ctl.arcam_av { + type arcam_av + port /dev/ttyS0 + } + +The "port" parameter is required and indicates the serial +port to be used to communicate with the amplifier. There is +an optional "zone" parameter, which accepts a value of +either "1" (default) or "2", that indicates which of the +amplifiers zones should be controlled. + +NB: You must ensure that any user accounts that are to use +this plugin have both read and write access rights for the +configured serial port. + +This plugin was developed and tested using an Arcam AVR 300 +amplifier. I believe most Arcam amplifiers use a sufficiently +similar control system to be compatible with this plugin but +your mileage may vary.

1 0

[alsa-devel] [PATCH 0/2] New Control Plugin - arcam-av
by Peter Stokes 03 Jan '09

03 Jan '09

Hi, I have developed a new ALSA control plugin that I would like to be considered for inclusion into the alsa-plugins package. The plugin is for exposing the controls of an Arcam AV amplifier attached to a PC via an RS-232 null-modem cable. These patches are against the release source for alsa-plugins-1.0.18. Best regards Peter Stokes

1 0

[alsa-devel] [PATCH - 1/2] ALSA: hda - add basic jack reporting functions to patch_conexant.c
by Ulrich Dangel 03 Jan '09

03 Jan '09

Added functions to report jack sense. As CXT5051_PORTB_EVENT has the same value as CONEXANT_MIC_EVENT two input devices for the microphone will be created if using CXT5051. Signed-off-by: Ulrich Dangel <uli(a)spamt.net> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 0760107..3cb640c 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -25,6 +25,8 @@ #include <linux/slab.h> #include <linux/pci.h> #include <sound/core.h> +#include <sound/jack.h> + #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" @@ -38,8 +40,21 @@ #define CONEXANT_HP_EVENT 0x37 #define CONEXANT_MIC_EVENT 0x38 +/* Conexant 5051 specific */ + +#define CXT5051_SPDIF_OUT 0x1C +#define CXT5051_PORTB_EVENT 0x38 +#define CXT5051_PORTC_EVENT 0x39 +struct conexant_jack { + + hda_nid_t nid; + int type; + struct snd_jack *jack; + +}; + struct conexant_spec { struct snd_kcontrol_new *mixers[5]; @@ -84,6 +99,9 @@ struct conexant_spec { unsigned int spdif_route; + /* jack detection */ + struct snd_array jacks; + /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; @@ -330,6 +348,86 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +static int conexant_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct conexant_spec *spec; + struct conexant_jack *jack; + const char *name; + + spec = codec->spec; + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; + + if (!jack) + return -ENOMEM; + + jack->nid = nid; + jack->type = type; + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +} + +static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct conexant_spec *spec = codec->spec; + struct conexant_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int present; + present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE; + + present = (present) ? jacks->type : 0 ; + + snd_jack_report(jacks->jack, + present); + } + jacks++; + } + } +} + +static int conexant_init_jacks(struct hda_codec *codec) +{ +#ifdef CONFIG_SND_JACK + struct conexant_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_init_verbs; i++) { + const struct hda_verb *hv; + + hv = spec->init_verbs[i]; + while (hv->nid) { + int err = 0; + switch (hv->param ^ AC_USRSP_EN) { + case CONEXANT_HP_EVENT: + err = conexant_add_jack(codec, hv->nid, + SND_JACK_HEADPHONE); + conexant_report_jack(codec, hv->nid); + break; + case CXT5051_PORTC_EVENT: + case CONEXANT_MIC_EVENT: + err = conexant_add_jack(codec, hv->nid, + SND_JACK_MICROPHONE); + conexant_report_jack(codec, hv->nid); + break; + } + if (err < 0) + return err; + ++hv; + } + } +#endif + return 0; + +} + static int conexant_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -342,6 +440,16 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { +#ifdef CONFIG_SND_JACK + struct conexant_spec *spec = codec->spec; + if (spec->jacks.list) { + struct conexant_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + snd_array_free(&spec->jacks); + } +#endif kfree(codec->spec); } @@ -1527,9 +1635,6 @@ static int patch_cxt5047(struct hda_codec *codec) /* Conexant 5051 specific */ static hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; -#define CXT5051_SPDIF_OUT 0x1C -#define CXT5051_PORTB_EVENT 0x38 -#define CXT5051_PORTC_EVENT 0x39 static struct hda_channel_mode cxt5051_modes[1] = { { 2, NULL }, -- 1.6.0.6

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[alsa-devel] [PATCH 0/4] ASoC updates for next
by Mark Brown 02 Jan '09

02 Jan '09

The following changes since commit 6a94cb73064c952255336cc57731904174b2c58f: Linus Torvalds (1): Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai If merging these patches just now is going to cause hassle then let me know. Mark Brown (1): ASoC: Clean up kerneldoc warnings Peter Ujfalusi (2): ASoC: TWL4030: Make the enum filter generic for twl4030 ASoC: TWL4030: DAPM based capture implementation Stephen Ware (1): ASoC: Fix pxa2xx-pcm checks for invalid DMA channels sound/soc/codecs/twl4030.c | 377 ++++++++++++++++++++++---------------------- sound/soc/codecs/twl4030.h | 7 + sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/soc-core.c | 46 +++--- sound/soc/soc-dapm.c | 10 +- 5 files changed, 227 insertions(+), 217 deletions(-)

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[alsa-devel] Subject: [PATCH 1/1] set invalid dma channel to -1 (and properly check for it) since 0 is not an invalid channel but -1 is.
by sware 01 Jan '09

01 Jan '09

Set the invalid dma channel to -1 (and check properly for it) in pxa2xx_pcm_hw_free(). Was assuming 0 is an invalid channel number but 0 is a valid pxa dma channel num. Signed-off-by: stephen <stephen.ware(a)eqware.net> --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index c670d08..53b9fb1 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -61,9 +61,9 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch) { + if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = 0; + prtd->dma_ch = -1; } return 0; -- 1.5.4.3

2 2

[alsa-devel] Control interconnection information API?
by Eliot Blennerhassett 01 Jan '09

01 Jan '09

New Year's Greetings, something that seems to be lacking in the ALSA API, or at least the way of describing controls (by name only), is any concept of the connectivity between the various controls and the codecs and output terminals. One use for such information is to build a GUI that shows this connectivity (instead of the current bucket'o'controls approach). Now there are a couple of special cases: * HDA driver exports the codec widget information via a proc file, allowing the codecgraph tool to visualise the connections. Widgets have numeric IDs and list the IDs of widgets that connect to their inputs. * ASoC DAPM (which I (not being an ASoC developer or user) have mostly been skimming over in this list). Reading this http://www.mjmwired.net/kernel/Documentation/sound/alsa/soc/dapm.txt 3. Codec Widget Interconnections. I can see that there are lists of connections. I'm not sure if these are visible to userspace via any API? So: Can either of these schemes be generalised to all ALSA drivers? Is there some other way to reveal the information (Connections TLV, new API etc?) Does anybody else thing this is worth pursuing, or is it just me? regards -- Eliot Blennerhassett AudioScience Inc

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Re: [alsa-devel] 2.6.28-rc5 broke ALSA headphone output for my Toughbooks.
by Takashi Iwai 01 Jan '09

01 Jan '09

At Tue, 23 Dec 2008 10:44:50 -0500, Kyler Laird wrote: > > My Panasonic CF-74 Toughbooks do not play audio through the headphone > jack starting with 2.6.28-rc5. The patch below is only to illustrate > what changes I made to restore headphone jack functionality. I do not > suggest using this patch as part of the mainline kernel because it > reverts some code that I'm sure is important to other users. > > Output of the ALSA script aadebug follows the patch. > > I'm continuing to investigate but I wanted to get this out in case the > solution is obvious to someone else. It would be great to have a fix in > 2.6.28. (2.6.28 is the first series to support all of the hardware in > the Toughbook.) Please run alsa-info.sh with --no-upload option and attach the generated files on both the headphone plugged and unplugged. Also, run preferably on both (old) working and non-working kernels to compare. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 11/13] sound/usb: use USB API functions rather than constants
by Takashi Iwai 01 Jan '09

01 Jan '09

At Mon, 29 Dec 2008 11:23:02 +0100 (CET), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > This set of patches introduces calls to the following set of functions: > > usb_endpoint_dir_in(epd) > usb_endpoint_dir_out(epd) > usb_endpoint_is_bulk_in(epd) > usb_endpoint_is_bulk_out(epd) > usb_endpoint_is_int_in(epd) > usb_endpoint_is_int_out(epd) > usb_endpoint_num(epd) > usb_endpoint_type(epd) > usb_endpoint_xfer_bulk(epd) > usb_endpoint_xfer_control(epd) > usb_endpoint_xfer_int(epd) > usb_endpoint_xfer_isoc(epd) > > In some cases, introducing one of these functions is not possible, and it > just replaces an explicit integer value by one of the following constants: > > USB_ENDPOINT_XFER_BULK > USB_ENDPOINT_XFER_CONTROL > USB_ENDPOINT_XFER_INT > USB_ENDPOINT_XFER_ISOC > > An extract of the semantic patch that makes these changes is as follows: > (http://www.emn.fr/x-info/coccinelle/) > > // <smpl> > @r1@ struct usb_endpoint_descriptor *epd; @@ > > - ((epd->bmAttributes & $USB_ENDPOINT_XFERTYPE_MASK\|3$) == > - $USB_ENDPOINT_XFER_CONTROL\|0$) > + usb_endpoint_xfer_control(epd) > > @r5@ struct usb_endpoint_descriptor *epd; @@ > > - ((epd->bEndpointAddress & $USB_ENDPOINT_DIR_MASK\|0x80$) == > - $USB_DIR_IN\|0x80$) > + usb_endpoint_dir_in(epd) > > @inc@ > @@ > > #include <linux/usb.h> > > @depends on !inc && (r1||r5)@ > @@ > > + #include <linux/usb.h> > #include <linux/usb/...> > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Applied now. Thanks. Takashi

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[alsa-devel] [PATCH] hda - fix name for ALC1200
by Wu Fengguang 01 Jan '09

01 Jan '09

On Thu, Dec 25, 2008 at 03:13:58PM +0200, Thomas Schneider wrote: > Hi, > > here is the output from alsa-info.sh > > http://www.alsa-project.org/db/?f=8282910dc807e1f141896886cd5f8fd574117000 > > I found another small mistake. In the Mixer my Soundcard is shown as "Realtek ALC888". > When i switch this to lines in patch_realtek.c then it is shown as "Realtek ALC1200". > > > static struct hda_codec_preset snd_hda_preset_realtek[] = { > ...... > > { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, > { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", > .patch = patch_alc883 }, > > ..... Good catch! Reversing the order should fix the name. Thanks, Fengguang --- hda - fix name for ALC1200 Move the more specific preset for ALC1200 above the general one for ALC888, so that it will have the chance to get matched and selected. Reported-by: Thomas Schneider <nailstudio(a)gmx.net> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -16638,9 +16638,9 @@ static struct hda_codec_preset snd_hda_p .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ };

2 1

[alsa-devel] [PATCH] ALSA: hda - Add HP Acacia detection
by Chris Bagwell 01 Jan '09

01 Jan '09

Add automatic mapping of HP Acacia motherboards to 3stack-hp. Allows for greater then 2 channel audio by enabling Channel Mode option in mixer. Motherboard specs: http://h10025.www1.hp.com/ewfrf/wc/document?docname=c01321559&lc=en&dlc=en&… Signed-off-by: Chris Bagwell <chris at cnpbagwell dot com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..410be84 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -8444,6 +8444,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),

2 1

Re: [alsa-devel] 2.6.28: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...
by Takashi Iwai 01 Jan '09

01 Jan '09

At Tue, 30 Dec 2008 07:28:14 -0500 (EST), Justin Piszcz wrote: > > On Tue, 30 Dec 2008, Justin Piszcz wrote: > > > The audio for this board works fine, there are no issues, but there are some > > cosmetic issues when the driver loads: > > > > 1. Unknown model. It's no bug. It's neither warning nor error message but a information message. > > 2. Wrong model number. They likely share the same ID and compatible. So don't worry about it. In short, nothing to fix, so far. thanks, Takashi -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Add DAPM based capture implementation
by Peter Ujfalusi 31 Dec '08

31 Dec '08

Hello, It has been a long time... Finally I have managed to get the DAPM based capture implementation working. I have not counted, but there were several dead end trials before I ended up with the current implementation. I have tested the capture implementation with the following configs: digimic0, Headset mic, AUX connected (not on the same board). So far the capture path behaved in a sane way, powering/enabling only the needed components in the capture path. All of the user controls are showing under the 'Capture' tab in alsamixer. Example for the board file maintainers to get the capture working: Let's say the board have the following input selection: Headset mic, AUXL/R (Line in), digimic0 static const struct snd_soc_dapm_widget BOARD_dapm_widgets[] = { SND_SOC_DAPM_MIC("Digital Mic", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_LINE("Line in", NULL); }; static const struct snd_soc_dapm_route audio_map[] = { {"DIGIMIC0", NULL, "Mic Bias 1"}, {"Mic Bias 1", NULL, "Digital Mic"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Mic"}, {"AUXL", NULL, "Line in"}, {"AUXR", NULL, "Line in"}, }; To record from the Headset mic: amixer sset 'Analog Left' 'Headset mic' # select the Headset for Analog left amixer sset 'TX1' 'Analog' # Select the Analog path for TX1 amixer sset 'TX2' 'Digimic1' # will disable the TX2 path, since digimic1 is # not connected amixer sset 'Analog' 2,2 # set the analog capture volume Have a Happy New Year! --- Peter Ujfalusi (2): ASoC: TWL4030: Make the enum filter generic for twl4030 ASoC: TWL4030: DAPM based capture implementation sound/soc/codecs/twl4030.c | 377 ++++++++++++++++++++++---------------------- sound/soc/codecs/twl4030.h | 7 + 2 files changed, 196 insertions(+), 188 deletions(-)

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[alsa-devel] Is Asoc suitable for my device?
by Xuesong Yang 31 Dec '08

31 Dec '08

hi all: I'm new to ALSA and still study it. My problem is I don't know if it suitable for my device use Asoc. In my platform, there is only a I2S bypass module, It has a route, select one of the I2S source as input,(from HDMI or cpu data) delay the data for some time and then output it to the backend. In theory, this device is very sample, but I find I can't get a easy way to implement it in ALSA, I'm lost in ALSA. Any suggestion will be greatly appreciated. Thanks.

2 2

[alsa-devel] problems with 2 outputs on snd-hda-intel
by Alex Ander 29 Dec '08

29 Dec '08

My configuration: http://www.alsa-project.org/db/?f=e33841b3aa560b031a3723ecf2e2c5040d2c1a16 Ever since I came back from the holidays I've been having this problem: My laptop has 3 jacks: 2 line out and 1 line in that can be switched to out. But, I can only get sound over the laptop-speakers and the line-in switched to out. The speaker stops playing when a jack is connected to one of the two outs, but I don't get any sound over these outputs. I have checked all mixer-settings in alsamixer, nothing works (except switching the mic to lineout, which doesn't fix the other 2 outs) As you can see in alsa-info I have forced a model-type to the module, I have found this on http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… I have tried all 4 modules listed at line 1015 (which should correspond to the SigmaTel STAC9228) of which 3stack seems to be the one set by default (with empty /etc/modprobe.conf) and is the only one that works with the mic-as-output option. I can give more info if you need it.

2 1

[alsa-devel] MIDI In on Audigy 2 ZS Live Drive
by Carl Bauer 28 Dec '08

28 Dec '08

On the Live Drive CD Bay thingy: MIDI in does not work. MIDI out (and everything else on the Live Drive) does work. For details, I have posted this problem before: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg23618.html http://forums.fedoraforum.org/showthread.php?t=198240 and seen it elsewhere: http://ubuntuforums.org/showthread.php?t=476633 I don't know if it's an ALSA bug or a kernel thing. My aadebug output is in the first of the above links. Or, my alsa-info.sh output is here: http://www.alsa-project.org/db/?f=5990a1a1b7476778a6cc82fff91997567db43a14 Thanks. Forgive me if this is the wrong list to post to. -Carl

1 0

[alsa-devel] [PATCH] alsa-info.sh - added extra checks
by Travis Place 28 Dec '08

28 Dec '08

Changelog: Added checks for Pulseaudio/Esound/aRts 'sound servers', Check if they are 1) installed 2) running A few small changes for the locale, and timezone - keep things more uniform. Bumped version to 0.4.53 Signed-off-by: Travis Place <wishie(a)wishie.net>

1 0

[alsa-devel] S/PDIF-Out on Asus M3A-H/HDMI with ALC1200
by Thomas Schneider 25 Dec '08

25 Dec '08

Hi, i am looking for a way to make Digital-Audio work with the external and internal S/PDIF-Output of my Motherboard Asus M3A-H/HDMI with ALC1200 Soundard. I use the latest unstable snapshot and i can´t make it work at the same time. On the external S/PDIF-Output is my Digital Receiver connected, on the internal one my GeForce 8500GT to get sound over HDMI on my LCT-TV. When i load the driver without any model-options my Soundcard is detected as Asus P5Q-EM HDMI ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:2716: hda_codec: model 'asus-p5q' is selected for config 1043:82fe (Asus P5Q-EM HDMI) [ 6.721700] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Headphone Playback Volume, skipped [ 6.721714] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Volume, skipped [ 6.721716] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Volume, skipped [ 6.721719] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Line-Out Playback Volume, skipped [ 6.721721] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave PCM Playback Volume, skipped [ 6.721729] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Switch, skipped [ 6.721731] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Switch, skipped Now i have sound on my LCD-TV over HDMI (internal S/PDIF-Out) When i load the driver with model=6stack-dig i have sound on my Digital-Receiver (external S/PDIF-Out). In the mailinglist-archive or in the doku i have seen, that the only difference between this models is the DIGOUT_NID asus-p5q = 0x10 6stack-dig = 0x06 I there a way to get both output to work at the same time so i have sound on my LCD-TV and my Digital-Receiver ? Regards Thomas

2 1

[alsa-devel] [PATCH 0/8]ALSA: ASoc: DaVinci: cleanup
by Troy Kisky 24 Dec '08

24 Dec '08

Sorry, for the long series to do something so simple, but there were unresolved objections when I first posted this to the davinci list. Hopefully, having this split into tiny pieces will make resolving those differences easier.

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[alsa-devel] [PATCH] ALSA: hda - Add missing terminators in patch_sigmatel.c
by Herton Ronaldo Krzesinski 24 Dec '08

24 Dec '08

Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 71d3a77..35b83dc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -841,6 +841,7 @@ static struct hda_verb stac92hd83xxx_core_init[] = { /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, + {} }; static struct hda_verb stac92hd71bxx_core_init[] = { @@ -850,6 +851,7 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} }; #define HD_DISABLE_PORTF 2 @@ -1681,6 +1683,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + {} /* terminator */ }; static unsigned int ref92hd71bxx_pin_configs[11] = {

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[alsa-devel] Urgent Help Needed:ALsa Querries
by SUBHRANIL CHOUDHURY 23 Dec '08

23 Dec '08

I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, Subhranil

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[alsa-devel] [PATCH] ASoC: Mark non-connected TWL4030 pins for pandora
by Grazvydas Ignotas 23 Dec '08

23 Dec '08

Pandora has all TWL4030 output pins floating, it uses external DAC for playback. Mark those outputs as not connected using DAPM calls. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index bd91594..0ffb3de 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -180,6 +180,19 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + snd_soc_dapm_nc_pin(codec, "OUTL"), + snd_soc_dapm_nc_pin(codec, "OUTR"), + snd_soc_dapm_nc_pin(codec, "EARPIECE"), + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"), + snd_soc_dapm_nc_pin(codec, "PREDRIVER"), + snd_soc_dapm_nc_pin(codec, "HSOL"), + snd_soc_dapm_nc_pin(codec, "HSOR"), + snd_soc_dapm_nc_pin(codec, "CARKITL"), + snd_soc_dapm_nc_pin(codec, "CARKITR"), + snd_soc_dapm_nc_pin(codec, "HFL"), + snd_soc_dapm_nc_pin(codec, "HFR"), + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); if (ret < 0) -- 1.5.4.3

2 1

[alsa-devel] [topic/asoc][PATCH] ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
by Jarkko Nikula 23 Dec '08

23 Dec '08

N810 bootloader muxes I2S pins for OMAP2420 EAC block while N810 ASoC drivers are using McBSP block so the kernel have to change configuration runtime. Author has not seen problems using kernel pin multiplexing on N810 but very many times unworking audio after forgotten to enable it and spending 15 minutes each time to figure it out again... This change makes it easier for other users as well. If problems arise, then they are better to find and fix in OMAP pin multiplexing framework. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a7b1d77..4f7f040 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -10,6 +10,7 @@ config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 select SND_OMAP_SOC_MCBSP + select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.5.6.5

2 1

[alsa-devel] [topic/asoc][PATCH]] ASoC: OMAP: Select OMAP pin multiplexing when using Nokia N810 ASoC drivers
by Jarkko Nikula 23 Dec '08

23 Dec '08

N810 bootloader muxes I2S pins for OMAP2420 EAC block while N810 ASoC drivers are using McBSP block so the kernel have to change configuration runtime. Author has not seen problems using kernel pin multiplexing on N810 but very many times unworking audio after forgotten to enable it and spending 15 minutes each time to figure it out again... This change makes it easier for other users as well. If problems arise, then they are better to find and fix in OMAP pin multiplexing framework. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a7b1d77..4f7f040 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -10,6 +10,7 @@ config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 select SND_OMAP_SOC_MCBSP + select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- 1.5.6.5

1 1

Re: [alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by Eliot Blennerhassett 22 Dec '08

22 Dec '08

SUBHRANIL CHOUDHURY wrote: > Eliot, > Suppose i need to change the DAC gain, ADC gain of the audio codec and > also disable and enable AGC of the codec i need to put some IOCTLs in > the driver. > So, in case of ALSA how do i implement those IOCTLs in the driver and > also expose them to the application through the ALSA library. No! For this kind of control You just need to implement standard alsa controls in your driver. BOOLEAN: AGC ON/OFF INTEGER: DAC gain. See "writing an ALSA driver" for details, and study the source of any ALSA driver. http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c843.htm -- Eliot

1 0

[alsa-devel] [topic/asoc][PATCH] ASoC: Fix DSP formats in SSM2602 audio codec
by Jarkko Nikula 22 Dec '08

22 Dec '08

Thanks to Troy Kisky <troy.kisky(a)boundarydevices.com> for noticing. - DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2 - DSP_B has 0-bit data delay which corresponds to submode 1 - Currently driver sets them opposite so swap the submode setting Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Cliff Cai <cliff.cai(a)analog.com> --- sound/soc/codecs/ssm2602.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2325aef..cac3736 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -454,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- 1.5.6.5

3 2

Re: [alsa-devel] [PATCH] ASoC: typo
by Liam Girdwood 22 Dec '08

22 Dec '08

On Mon, 2008-12-22 at 17:40 +0100, Roel Kluin wrote: > Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> > --- > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index acd68ef..227c865 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, > prtd->dma_data = dma_data; > err = omap_request_dma(dma_data->dma_req, dma_data->name, > omap_pcm_dma_irq, substream, &prtd->dma_ch); > - if (!err & !cpu_is_omap1510()) { > + if (!err && !cpu_is_omap1510()) { > /* > * Link channel with itself so DMA doesn't need any > * reprogramming while looping the buffer Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Liam

2 1

[alsa-devel] ASoC updaets
by Mark Brown 22 Dec '08

22 Dec '08

The following changes since commit a31501d1041c9d0a6c3f520736ae2b2fa081493a: Takashi Iwai (1): ALSA: ASoC - Add missing __devexit annotation to wm8350.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Jarkko Nikula (2): ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers ASoC: Fix DSP formats in SSM2602 audio codec sound/soc/codecs/ssm2602.c | 4 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/osk5912.c | 4 ++-- 4 files changed, 7 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [topic/asoc][PATCH 0/1] ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
by Jarkko Nikula 22 Dec '08

22 Dec '08

Hi This fix is generated against topic/asoc and don't apply against mainline. IMO this can go to next merge window with preceding patches since OSK5912 in mainline should work, just format is defined wrong and all new drivers where this affects should anyway go to the top of topic/asoc. -- Jarkko

3 3

[alsa-devel] patch 1/1: ASoC, DMA crash with application- resolved
by naveenkrishna.ch 22 Dec '08

22 Dec '08

Hi ALSA SOC, I am using a OMAP3 + TWL4030 combination with ASoC in the sound/soc/omap/omap-pcm.c DMA channels are being freed after unlinking in the omap_pcm_hw_free function Which is causing a crash during applications So, stopping the DMA channels before unlinking resolves the issue. here is the patch that solves the issue diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e9084fd..679e384 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -116,6 +116,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_data == NULL) return 0; + omap_stop_dma(prtd->dma_ch); if (!cpu_is_omap1510()) omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); omap_free_dma(prtd->dma_ch); Is this ok or is there any better way out -- Thanks, (: Naveen Krishna Ch :)

4 7

Re: [alsa-devel] [PATCH 0/8]ALSA: ASoc: DaVinci: cleanup
by David Brownell 22 Dec '08

22 Dec '08

On Sunday 21 December 2008, Rajashekhara, Sudhakar wrote: > the existing meaning is correct w.r.t davinci McBSP. Not entirely. DaVinci doesn't have McBSP; it has ASP. Having all this code refer to a McBSP -- as found for example on OMAP chips -- is confusing and incorrect. - Dave

1 0

[alsa-devel] Sigmatel codec on Dell Vostro 1400
by Paulo Cavalcanti 21 Dec '08

21 Dec '08

Hi, I have a Dell Vostro 1400 laptop, which uses the sigmatel codec STAC9228. This laptop has a digital built in mic and a connector for an external analog mic. It also has two headphones connectors. Using yesterday's alsa tarball model=ref or ref-no-jd - sound in headphones only, external analog mic works. No digital mic. model=3stack,5stack - no sound at all. model=dell-3stack - sound works for headphones and speakers (jack detection mutes speakers when headphones are inserted), no analog external mic, and digital mic works, but sound is really too low. Therefore, I would be very pleased if I could either use the analog mic with model=dell-3stack, or if the built in digital mic had a louder sound. Using hda-verb-0.3, the pins are: 0xa - headphones (first connector - the one which works) 0xf - headphones (second connector - mute) 0xe - analog mic The information from alsa-info is here: http://www.alsa-project.org/db/?f=6a64faf3cd93c8c805e868c33eff88aec29508c8 Thanks. -- Paulo Roma Cavalcanti LCG - UFRJ

1 0

[alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by SUBHRANIL CHOUDHURY 21 Dec '08

21 Dec '08

Dear all, I am new to ALSA programming. Suppose if i have to add some hardware specific feature in the ALSA sound driver and expose to the application via the ALSA library, how can do that? Please reply if you have some inputs. Anything is welcome. Thanks in advance. Regards, Subhranil

2 1

[alsa-devel] [PATCH] hda: fix incorrect mixer index values for 92hd83xx
by Matthew Ranostay 21 Dec '08

21 Dec '08

Fixed incorrect mixer index values for 92hd83xx codec's audio input mixer. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 70181d5..efa1fb0 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1101,21 +1101,21 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT), - HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT), /* - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT), */ { } /* end */ };

2 2

[alsa-devel] [PATCH] hda: dinput_mux check
by Matthew Ranostay 21 Dec '08

21 Dec '08

Add check to determine if dinput_mux is set by any of patch_stac*() functions, otherwise a invalid pointer my be referenced causing gibberish to mixer values. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 70181d5..fa87893 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3601,7 +3601,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctls.list; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux;

2 2

[alsa-devel] somebody ever see ALSA applied in robotics ?
by Ronald 20 Dec '08

20 Dec '08

Hi, all, I'm trying to develop a system which can determine the direction(s) sound comes from. In the end, it will be put in an R/c submarine, on a gumstix linux board (see http://www.subcommittee.com for my hobby) But for now, I develop in air, with ubuntu. all I want is to record using both the mic channel and the line in (where another 2 mics are attached with a small preamp), and I want to record while emitting a chirp, analyzing echo's. the concept has been shown to work. see for instance http://eddiem.com/projects/chirp/chirp.htm I am assuming that since all this goes via the same sound card, the same crystal is used, and samples from mic and line in are essentially synchronized (I only need it to be sync-ish for < 0,1 sec..) Now I have read a lot of ALSA tutorials and such like, but I'm lost. a lot is about humans consuming nice harmonics, sharing and mixing, which is not applicable to my case. I have the nagging feeling OSS is a better fit for my case, but OSS is apparently on a 60 day licence now. And, I am a newbie coder. no surprise there ;-) So, these are my questions : - Do I need to combine mic and line-in into one device using the config file ? do I get a 3 channel device then ? does a plug-in add latency ? - can I sync the transmission of a short chirp with the start of the recording on these 3 channels ? is starting them close after each other in the program OK, or is there possibly large latency caused *after* I call the API ? the "real" sync'ing will be the (first) recording of the chirp once it leaves the speaker, so it isn't critical, but I would like to know because it might save a few bytes useless samples. - Does anybody out there have any links, code snippets or general published info on how to apply ALSA to the robotics field ? Ps I have read the el-cheapo hack with multiple soundcards, and while a very good idea, I cannot use it in the boat later on.. thanks for any input you might have. Regards, Ronald van Aalst

3 2

[alsa-devel] Unknown symbols in snd-fm801.ko
by Ozan Çağlayan 20 Dec '08

20 Dec '08

Hi, Yesterday I was trying to compile the latest snapshot of alsa-driver on top of kernel 2.6.28_rc8. When looking through the console output of make, I saw two warnings given by LD reporting two unknown symbols in snd-fm801.ko(driver for a ForteMedia sound card): Building modules, stage 2. MODPOST 170 modules WARNING: "snd_tea575x_init" [/var/pisi/alsa-driver-1.0.18a_20081214-39/work/alsa-driver/pci/snd-fm801.ko] undefined! WARNING: "snd_tea575x_exit" [/var/pisi/alsa-driver-1.0.18a_20081214-39/work/alsa-driver/pci/snd-fm801.ko] undefined! laptop alsa-driver # grep -rHIn "snd_tea575x_init" * alsa-kernel/i2c/other/tea575x-tuner.c:195:void snd_tea575x_init(struct snd_tea575x *tea) alsa-kernel/i2c/other/tea575x-tuner.c:251:EXPORT_SYMBOL(snd_tea575x_init); alsa-kernel/include/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); alsa-kernel/pci/fm801.c:1447: snd_tea575x_init(&chip->tea); i2c/other/tea575x-tuner.c:236:void snd_tea575x_init(struct snd_tea575x *tea) i2c/other/tea575x-tuner.c:309:EXPORT_SYMBOL(snd_tea575x_init); include/sound/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); pci/fm801.c:1448: snd_tea575x_init(&chip->tea); pci/fm801.c.orig:1447: snd_tea575x_init(&chip->tea); sound/i2c/other/tea575x-tuner.c:195:void snd_tea575x_init(struct snd_tea575x *tea) sound/i2c/other/tea575x-tuner.c:251:EXPORT_SYMBOL(snd_tea575x_init); sound/include/tea575x-tuner.h:51:void snd_tea575x_init(struct snd_tea575x *tea); sound/pci/fm801.c:1447: snd_tea575x_init(&chip->tea); So this symbols are exported by tea575x-tuner.c. There's a card option parameter in configuration of alsa-driver called 'fm801-tea575x-bool'.But eventhough, I'm configuring the package with, "--with-card-options=all", I'm getting this warnings. As a consequence it's impossible to load the module snd-fm801. I think that it's a problem with Kconfig stuff but I really don't get how alsa-driver package manages the whole thing with alsa-kernel directory, patchsets etc.(BTW, is there any documentation about the mechanism in alsa-driver, and the functions of separate git repos alsa-driver, alsa-kernel, stub, etc.?) I'm having the issue with 2.6.25.x && (1.0.17 || 1.0.18), 2.6.28_rc8 && (1.0.18) so the problem may not be caught until this time because there exists a few distributions which ships alsa-driver(and other kernel modules) separately. Regards, -- Ozan Çağlayan <ozan_at_pardus.org.tr>

2 6

[alsa-devel] ice1712 error
by Udo van den Heuvel 20 Dec '08

20 Dec '08

Hello, I found out the ICE1712 card is sometimes detected OK and sometimes isn't detected correctly. Cold boot/warm boot doesn't really matter. I tried kernel 2.6.27.9 and older. Card is is fine when detected. Behaviour doesn't change when I try a different PCI slot. When the card fails: # dmesg|grep 01:09 PCI: 0000:01:09.0 reg 10 io port: [a000, a01f] PCI: 0000:01:09.0 reg 14 io port: [a400, a40f] PCI: 0000:01:09.0 reg 18 io port: [a800, a80f] PCI: 0000:01:09.0 reg 1c io port: [ac00, ac3f] pci 0000:01:09.0: supports D2 ICE1712: probe of 0000:01:09.0 failed with error -12 When the card is OK: PCI: 0000:01:09.0 reg 10 io port: [a000, a01f] PCI: 0000:01:09.0 reg 14 io port: [a400, a40f] PCI: 0000:01:09.0 reg 18 io port: [a800, a80f] PCI: 0000:01:09.0 reg 1c io port: [ac00, ac3f] pci 0000:01:09.0: supports D2 ICE1712 0000:01:09.0: PCI INT A -> Link[APC4] -> GSI 19 (level, low) -> IRQ 19 I booted into single user mode each time and checked /proc/asound/cards after each boot and then looked at dmesg. Does this mean the card is toast? What does the -12 mean? What should I try next? Kind regards, Udo

3 13

[alsa-devel] [PATCHv2 1/1] Fix a Oops bug in omap soc driver.
by Stanley.Miao 20 Dec '08

20 Dec '08

There will be a Oops or frequent underrun messages when playing music with omap soc driver, this is because a data region is incorretly sized, other data region will be overwriten when writing to this data region. Signed-off-by: Stanley.Miao <stanley.miao(a)windriver.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index ce580a9..2adfbce 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -233,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) goto out; - prtd = kzalloc(sizeof(prtd), GFP_KERNEL); + prtd = kzalloc(sizeof(struct omap_runtime_data), GFP_KERNEL); if (prtd == NULL) { ret = -ENOMEM; goto out; -- 1.5.6.3

6 6

[alsa-devel] ALsa Querries about buffer size configuration
by SUBHRANIL CHOUDHURY 20 Dec '08

20 Dec '08

I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, Subhranil

2 1

[alsa-devel] Additing a new Hardware Specific feature in the ALSA driver
by SUBHRANIL CHOUDHURY 20 Dec '08

20 Dec '08

On Sat, Dec 20, 2008 at 7:41 PM, SUBHRANIL CHOUDHURY <subhra85(a)gmail.com>wrote: > > > Dear all, > I am new to ALSA programming. > Suppose if i have to add some hardware specific feature in the ALSA sound > driver and expose to the application via the ALSA library, how can do that? > > Please reply if you have some inputs. > Anything is welcome. > Thanks in advance. > > Regards, > Subhranil

1 0

Re: [alsa-devel] [PATCH 01/01] alsa: sound/soc/pxa/pxa2xx_pcm.c, check and set invalid dma_ch to be <= to 0 in pxa2xx_pcm_hw_free()
by Mark Brown 20 Dec '08

20 Dec '08

On Fri, Dec 19, 2008 at 09:52:22PM -0800, sware wrote: > From: Stephen Ware <stephen.ware(a)eqware.net>, Dennis O'Brien <dennis.obrien(a)eqware.net> ... > Signed-off-by: Dennis O'Brien <dennis.obrien(a)eqware.net> The patch itself is good but could you please clarify what's going on with the author and signoff lines (apart from anything else the multiple authors appears to be confusing git, it's ignoring Dennis). The combination of git format-patch and send-email is handy for generating something that applies easily.

1 0

[alsa-devel] [PATCH 1/4] ALSA: ASoC: tlv320aic3x add dsp_a
by Troy Kisky 20 Dec '08

20 Dec '08

Add SND_SOC_DAIFMT_DSP_A mode option. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b76bcc3..b9b67ed 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,6 +847,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; u8 iface_areg, iface_breg; + int delay = 0; iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -872,6 +873,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, SND_SOC_DAIFMT_INV_MASK)) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -888,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set iface */ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195a..40a6c2a 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -35,6 +35,8 @@ #define AIC3X_ASD_INTF_CTRLA 8 /* Audio serial data interface control register B */ #define AIC3X_ASD_INTF_CTRLB 9 +/* Audio serial data interface control register C */ +#define AIC3X_ASD_INTF_CTRLC 10 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ -- 1.5.4.3

2 4

[alsa-devel] "No codecs found" on Intel HDA + AD1988A
by Alexander Chemeris 19 Dec '08

19 Dec '08

Hi, I have Asus P5B-V motherboard with integrated Intel HDA audio and AD1988A codec (I see the chip on the board). About half a year ago or so it used to work, but now when snd-hda-intel module is loaded it claims it can't find a codec: $ dmesg | grep -i hda [ 13.648205] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 13.648236] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 13.680537] hda-intel: no codecs found! [ 13.680589] HDA Intel 0000:00:1b.0: PCI INT A disabled I tried different recent versions of ALSA to no avail. So far I've tried: * 1.0.17 with Ubuntu patches * 1.0.18 with Ubuntu patches * 1.0.18a downloaded half an hour ago from here: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… I also tried to compile 1.0.16, but seems it is not compatible with the kernel: $ uname -a Linux nidus 2.6.27-10-generic #1 SMP Fri Nov 21 19:19:18 UTC 2008 x86_64 GNU/Linux When I tried 1.0.18a, I also manually loaded snd-hda-codec_analog before loading snd-hda-intel without any success. And yeah, when I just run `modprobe snd-hda-intel` only following modules are loaded: $ lsmod | grep snd snd_hda_intel 37864 0 snd_hda_codec 86784 1 snd_hda_intel snd_hwdep 17160 1 snd_hda_codec snd_pcm_oss 52256 0 snd_mixer_oss 25088 1 snd_pcm_oss snd_pcm 98824 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_oss 43776 0 snd_seq_midi_event 16768 1 snd_seq_oss snd_seq 67744 4 snd_seq_oss,snd_seq_midi_event snd_timer 33424 2 snd_pcm,snd_seq snd_seq_device 16788 2 snd_seq_oss,snd_seq snd 83144 10 snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device soundcore 16800 1 snd snd_page_alloc 18576 2 snd_hda_intel,snd_pcm What can I do to help get my audio working again? I don't want to install different kernel or restart computer (well, as a last resort, may be...), but I easily can test different any patches, etc. PS I don't think it matters, but I use Ubuntu 8.10 x64. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000

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Re: [alsa-devel] No HDMI audio on Asus P5Q-EM
by Chris Pimlott 19 Dec '08

19 Dec '08

Forgot to join the list first, also here is the bugtrack issue: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4281 On Thu, Dec 11, 2008 at 3:41 PM, Chris Pimlott <pimlottc(a)gmail.com> wrote: > I have an Asus P5Q-EM motherboard, with Intel G45 chipset and Realtek ALC > 1200 integrated audio, with HDMI output. (Specifications: > http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…)<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…>[ > ^<http://www.asus.com/products.aspx?modelmenu=2&model=2413&l1=3&l2=11&l3=761&…> > ] > > The analog output has always worked fine (from the default Mythbuntu > install), but I do not get any output when trying to use the HDMI. I am > testing with aplay: "aplay -D plughw:0,3 > /usr/share/sounds/alsa/Front_Center.wav". plughw:0,1 with the same command > works (on analog out). > > I see three devices, "IEC958", "IEC958 Default PCM" and "IEC958 1", which > are all unmuted. I have tried flipped the copyright setting with "iecset -n > 1 copy on" but that has made no difference. > > I have tried a number of versions, including the stock 2.6.27 kernel > package shipped in Mythbuntu, alsa-1.0.18 and alsa-1.0.18a, as well as > version of the sound-2.6 from 2008-11-23 and today. The most recent I'm > trying is sound-2.6 version is b13a9d80db0f1f4ace2863e8824a3d3299730d0a. > > I've notice there's been a fair amount of updates to this area of the > snd_hda_intel driver, and some changes to how the device has shown up in > recent releases. Should I even expect this to work, or is it still > "in-progress"? Thanks. > > Chris Pimlott >

2 1

[alsa-devel] arecord ..xrun!!!!
by vasudha rao 19 Dec '08

19 Dec '08

Hi .. I am trying to record .wav files.. I did like, $arecord -D copy test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono xrun!!! (at least 0.251 ms long) xrun!!! (at least 0.071 ms long) xrun!!! (at least 0.072 ms long) . . . When I run this test.wav, no sound!!!! Am I doing wrong anywhere?? Thanks:-)

3 2

[alsa-devel] arecord working now..
by vasudha rao 19 Dec '08

19 Dec '08

Hi.. I am able to record now.. I used arecord like, $arecord -d 10 -f cd -t wav -D copy foobar.wav and I play like, $aplay -D hw:0,0 foobar.wav All works fine..:-) But the thing is I dont know why it was not working and why it is working now..?? Please tel me if u found it why??and if time permits for u.. Thank you ..

1 0

Re: [alsa-devel] Patch to /sound/soc/pxa/pxa2xx_pcm.c, WHO IS INTERESTED?
by Mark Brown 19 Dec '08

19 Dec '08

On Thu, Dec 18, 2008 at 10:34:53AM -0800, sware wrote: > Patch is to 2.6.27.9 (yesterday), notice .10 is up today, > patch should still apply. Thanks for your patch. Unfortunately all this code has been moved about a lot in the latest code in order to share the code with the non-ASoC AC97 driver so your patch won't apply - see the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for the latest code. Could you rebase your work against that, please? Also, please include a Signed-off-by line for patch submissions - for details see Documentation/SubmittingPatches > diff -uprN -X linux-2.6.27.9.orig/Documentation/dontdiff linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c > --- linux-2.6.27.9.orig/sound/soc/pxa/pxa2xx-pcm.c 2008-12-13 15:56:16.000000000 -0800 > +++ linux-2.6.27.9.snd_modified/sound/soc/pxa/pxa2xx-pcm.c 2008-12-17 14:09:02.000000000 -0800 > @@ -137,13 +137,18 @@ static int pxa2xx_pcm_hw_free(struct snd > { > struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; > > - if (prtd && prtd->params) > + if (!prtd) > + return 0; > + if (!prtd->params) > + return 0; > + prtd should always be there - the code should at least complain loudly if it's NULL. params might potentially be hit, though. > *prtd->params->drcmr = 0; > > - if (prtd->dma_ch) { > + if (prtd->dma_ch >= 0) { > snd_pcm_set_runtime_buffer(substream, NULL); > pxa_free_dma(prtd->dma_ch); > - prtd->dma_ch = 0; > + prtd->dma_ch = -1; /* invalid channel */ > + prtd->params = NULL; > } This bit is good. > @@ -266,6 +271,8 @@ static int pxa2xx_pcm_close(struct snd_p > dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, > prtd->dma_desc_array, prtd->dma_desc_array_phys); > kfree(prtd); > + runtime->private_data = NULL; > + > return 0; > } As is this.

2 2

[alsa-devel] Streams
by James Courtier-Dutton 19 Dec '08

19 Dec '08

Hi, There was some talk some time ago of maybe moving alsa to a streams approach instead of ring buffer and periods approach. Has anything become of this? Is there a plan to improve the abstraction between hardware and user space regarding constraints on period sizes when doing sample rate conversion? Kind Regards James

2 1

[alsa-devel] 44.1khz capture on CA0106?
by Takashi Iwai 19 Dec '08

19 Dec '08

Hi, can anyone confirm that 44.1kHz capture on CA0106 causes the problem with 48kHz playback? Typically, recording via 44.kHz using hw device, then run 48kHz playback. % arecord -fcd -Dhw:0 foo.wav % aplay -Dhw:0 48khz-samples.wav I got a bug report about this, and disabling 44.1kHz capture fixes the problem. His case is with MSI K8N mobo, and I'd like to know whether this happens in general on CA0106. The bug report is found in https://bugzilla.novell.com/show_bug.cgi?id=447624 thanks, Takashi

1 0

[alsa-devel] ca0106 SPDIF defaults
by Ben Stanley 19 Dec '08

19 Dec '08

Hi, I previously reported problems that are still plaguing me [1] to do with SPDIF settings that don't reset properly. I am now looking at doing a patch to the ca0106 driver to fix this problem. It appears I have to add a new control to be able to set the default settings. I presume this introduces complications with existing settings in asound.state? Is there some collected wisdom on how to add new controls while providing a migration/upgrade/interoperability path? Ben Stanley. [1] http://article.gmane.org/gmane.linux.alsa.devel/55406/match=independent+pcm…

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 19 Dec '08

19 Dec '08

The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583: Stanley.Miao (1): ASoC: TWL4030: hands-free start-up sequence. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Alexander Beregalov (1): ASoC: switch davinci DPRINTK to pr_debug() Mark Brown (3): ASoC: Ease merge difficulties from new architectures ASoC: Complain if we fail to create DAPM controls ASoC: Add WM8350 AudioPlus codec driver include/linux/mfd/wm8350/audio.h | 38 +- sound/soc/Kconfig | 10 +- sound/soc/Makefile | 12 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8350.c | 1583 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 20 + sound/soc/davinci/davinci-pcm.c | 18 +- sound/soc/soc-dapm.c | 6 +- 9 files changed, 1669 insertions(+), 24 deletions(-) create mode 100644 sound/soc/codecs/wm8350.c create mode 100644 sound/soc/codecs/wm8350.h

3 6

[alsa-devel] [ASoC] TWL4030 underrun error
by naveenkrishna.ch 18 Dec '08

18 Dec '08

Hello, I am working on OMAP3 + TWL4030 combination for ASoC codec The omap generic omap-pcm.c file has static const struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_96000, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, }; following structure, When i play with aplay XXX.wav Rare but i used to get some underrun messages i found thses message are coming from aplay. underrun!!! (at least 10271.545 ms long) underrun!!! (at least 904.663 ms long) underrun!!! (at least 771.790 ms long) underrun!!! (at least 3393.433 ms long) underrun!!! (at least 777.314 ms long) I dint found much info about this in net too. How can i solve this, can any one help me out -- Thanks, (: Naveen Krishna Ch :)

3 2

[alsa-devel] [PATCH] ALSA: Add support for TV-out to the jack reporting API
by Jani Nikula 18 Dec '08

18 Dec '08

Add support for reporting new jack types SND_JACK_TVOUT and SND_JACK_AVOUT (a combination of LINEOUT and TVOUT) to the jack reporting API. Also add the corresponding SW_TVOUT_INSERT switch to the input system header. Signed-off-by: Jani Nikula <ext-jani.1.nikula(a)nokia.com> --- include/linux/input.h | 1 + include/sound/jack.h | 2 ++ sound/core/jack.c | 6 ++++++ 3 files changed, 9 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index abd223b..ea5b224 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -646,6 +646,7 @@ struct input_absinfo { #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ +#define SW_TVOUT_INSERT 0x08 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 2e0315c..69fc1f0 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -37,6 +37,8 @@ enum snd_jack_types { SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_TVOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_TVOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 7ca905c..23c3f3e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -105,6 +105,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_LINEOUT) input_set_capability(jack->input_dev, EV_SW, SW_LINEOUT_INSERT); + if (type & SND_JACK_TVOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_TVOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_LINEOUT) input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, status & SND_JACK_LINEOUT); + if (jack->type & SND_JACK_TVOUT) + input_report_switch(jack->input_dev, SW_TVOUT_INSERT, + status & SND_JACK_TVOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); -- 1.6.0.4

3 8

[alsa-devel] aplay..no sound..no error..
by vasudha rao 17 Dec '08

17 Dec '08

Hi all.. I have installed alsa for my embedded board..(at91sam9263) when I use aplay utility I got, $ aplay a.wav Playing WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono I am not getting any error message.. But no sound from speaker also.. What else I can try to listen sound from speaker..?? Thanks,

3 2

[alsa-devel] SND_SOC_DAIFMT_NB_NF meaning
by Troy Kisky 17 Dec '08

17 Dec '08

A quick question here. Does SND_SOC_DAIFMT_NB_NF have a standard meaning or does it depend on what mode your in (I2S, DSP_A, right just). From looking at the sources, SND_SOC_DAIFMT_NB_NF usually means that data is valid on the rising BCLK and start of frame is rising edge if DSP mode. The reason I ask, is that I'd like to pass the same flags to the codec and the cpu_dai and have them agree with the interpretation. Thanks Troy

2 1

Re: [alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Takashi Iwai 17 Dec '08

17 Dec '08

At Wed, 17 Dec 2008 10:49:08 +0100, Klaus Schulz wrote: > > Takashi. > > As you can see below SND_VERBOSE_PROCFS is y. But DEBUG and VERBOSE_PRINTK is > NO! These are irrelevant. > BTW: I am running a 2.6.28-rc6-zen1 kernel. I don't know what it means. But the stream* proc file should be created for snd-usb-audio when CONFIG_PROC_FS and CONFIG_SND_VERBOSE_PROCFS are set. At least, there is no change in the ALSA upstream side. Takashi > > Cheers > Klaus > > CONFIG_SOUND=m > CONFIG_SOUND_OSS_CORE=y > CONFIG_SND=m > CONFIG_SND_TIMER=m > CONFIG_SND_PCM=m > CONFIG_SND_HWDEP=m > CONFIG_SND_RAWMIDI=m > CONFIG_SND_SEQUENCER=m > CONFIG_SND_SEQ_DUMMY=m > CONFIG_SND_OSSEMUL=y > CONFIG_SND_MIXER_OSS=m > CONFIG_SND_PCM_OSS=m > CONFIG_SND_PCM_OSS_PLUGINS=y > CONFIG_SND_SEQUENCER_OSS=y > CONFIG_SND_DYNAMIC_MINORS=y > CONFIG_SND_SUPPORT_OLD_API=y > CONFIG_SND_VERBOSE_PROCFS=y > # CONFIG_SND_VERBOSE_PRINTK is not set > # CONFIG_SND_DEBUG is not set > CONFIG_SND_VMASTER=y > CONFIG_SND_MPU401_UART=m > CONFIG_SND_OPL3_LIB=m > CONFIG_SND_OPL4_LIB=m > CONFIG_SND_VX_LIB=m > CONFIG_SND_AC97_CODEC=m > CONFIG_SND_DRIVERS=y > CONFIG_SND_PCSP=m > CONFIG_SND_DUMMY=m > CONFIG_SND_VIRMIDI=m > CONFIG_SND_MTPAV=m > CONFIG_SND_MTS64=m > CONFIG_SND_SERIAL_U16550=m > CONFIG_SND_MPU401=m > CONFIG_SND_PORTMAN2X4=m > CONFIG_SND_AC97_POWER_SAVE=y > CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0 > CONFIG_SND_WSS_LIB=m > CONFIG_SND_SB_COMMON=m > CONFIG_SND_SB8_DSP=m > CONFIG_SND_SB16_DSP=m > CONFIG_SND_ISA=y > CONFIG_SND_ADLIB=m > CONFIG_SND_AD1816A=m > CONFIG_SND_AD1848=m > CONFIG_SND_ALS100=m > CONFIG_SND_AZT2320=m > CONFIG_SND_CMI8330=m > CONFIG_SND_CS4231=m > CONFIG_SND_CS4232=m > CONFIG_SND_CS4236=m > CONFIG_SND_DT019X=m > CONFIG_SND_ES968=m > CONFIG_SND_ES1688=m > CONFIG_SND_ES18XX=m > CONFIG_SND_SC6000=m > CONFIG_SND_GUSCLASSIC=m > CONFIG_SND_GUSEXTREME=m > CONFIG_SND_GUSMAX=m > CONFIG_SND_INTERWAVE=m > CONFIG_SND_INTERWAVE_STB=m > CONFIG_SND_OPL3SA2=m > CONFIG_SND_OPTI92X_AD1848=m > CONFIG_SND_OPTI92X_CS4231=m > CONFIG_SND_OPTI93X=m > CONFIG_SND_MIRO=m > CONFIG_SND_SB8=m > CONFIG_SND_SB16=m > CONFIG_SND_SBAWE=m > CONFIG_SND_SB16_CSP=y > CONFIG_SND_SGALAXY=m > CONFIG_SND_SSCAPE=m > CONFIG_SND_WAVEFRONT=m > CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL=y > CONFIG_SND_PCI=y > CONFIG_SND_AD1889=m > CONFIG_SND_ALS300=m > CONFIG_SND_ALS4000=m > CONFIG_SND_ALI5451=m > CONFIG_SND_ATIIXP=m > CONFIG_SND_ATIIXP_MODEM=m > CONFIG_SND_AU8810=m > CONFIG_SND_AU8820=m > CONFIG_SND_AU8830=m > CONFIG_SND_AW2=m > CONFIG_SND_AZT3328=m > CONFIG_SND_BT87X=m > # CONFIG_SND_BT87X_OVERCLOCK is not set > CONFIG_SND_CA0106=m > CONFIG_SND_CMIPCI=m > CONFIG_SND_OXYGEN_LIB=m > CONFIG_SND_OXYGEN=m > CONFIG_SND_CS4281=m > CONFIG_SND_CS46XX=m > CONFIG_SND_CS46XX_NEW_DSP=y > CONFIG_SND_CS5530=m > CONFIG_SND_CS5535AUDIO=m > CONFIG_SND_DARLA20=m > CONFIG_SND_GINA20=m > CONFIG_SND_LAYLA20=m > CONFIG_SND_DARLA24=m > CONFIG_SND_GINA24=m > CONFIG_SND_LAYLA24=m > CONFIG_SND_MONA=m > CONFIG_SND_MIA=m > CONFIG_SND_ECHO3G=m > CONFIG_SND_INDIGO=m > CONFIG_SND_INDIGOIO=m > CONFIG_SND_INDIGODJ=m > CONFIG_SND_EMU10K1=m > CONFIG_SND_EMU10K1X=m > CONFIG_SND_ENS1370=m > CONFIG_SND_ENS1371=m > CONFIG_SND_ES1938=m > CONFIG_SND_ES1968=m > CONFIG_SND_FM801=m > CONFIG_SND_FM801_TEA575X_BOOL=y > CONFIG_SND_FM801_TEA575X=m > CONFIG_SND_HDA_INTEL=m > # CONFIG_SND_HDA_HWDEP is not set > # CONFIG_SND_HDA_INPUT_BEEP is not set > CONFIG_SND_HDA_CODEC_REALTEK=y > CONFIG_SND_HDA_CODEC_ANALOG=y > CONFIG_SND_HDA_CODEC_SIGMATEL=y > CONFIG_SND_HDA_CODEC_VIA=y > CONFIG_SND_HDA_CODEC_ATIHDMI=y > CONFIG_SND_HDA_CODEC_NVHDMI=y > CONFIG_SND_HDA_CODEC_CONEXANT=y > CONFIG_SND_HDA_CODEC_CMEDIA=y > CONFIG_SND_HDA_CODEC_SI3054=y > CONFIG_SND_HDA_GENERIC=y > CONFIG_SND_HDA_POWER_SAVE=y > CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0 > CONFIG_SND_HDSP=m > CONFIG_SND_HDSPM=m > CONFIG_SND_HIFIER=m > CONFIG_SND_ICE1712=m > CONFIG_SND_ICE1724=m > CONFIG_SND_INTEL8X0=m > CONFIG_SND_INTEL8X0M=m > CONFIG_SND_KORG1212=m > CONFIG_SND_MAESTRO3=m > CONFIG_SND_MIXART=m > CONFIG_SND_NM256=m > CONFIG_SND_PCXHR=m > CONFIG_SND_RIPTIDE=m > CONFIG_SND_RME32=m > CONFIG_SND_RME96=m > CONFIG_SND_RME9652=m > CONFIG_SND_SIS7019=m > CONFIG_SND_SONICVIBES=m > CONFIG_SND_TRIDENT=m > CONFIG_SND_VIA82XX=m > CONFIG_SND_VIA82XX_MODEM=m > CONFIG_SND_VIRTUOSO=m > CONFIG_SND_VX222=m > CONFIG_SND_YMFPCI=m > CONFIG_SND_SPI=y > CONFIG_SND_USB=y > CONFIG_SND_USB_AUDIO=m > CONFIG_SND_USB_USX2Y=m > CONFIG_SND_USB_CAIAQ=m > CONFIG_SND_USB_CAIAQ_INPUT=y > # CONFIG_SND_USB_US122L is not set > CONFIG_SND_PCMCIA=y > CONFIG_SND_VXPOCKET=m > CONFIG_SND_PDAUDIOCF=m > CONFIG_SND_SOC=m > # CONFIG_SND_SOC_ALL_CODECS is not set > CONFIG_SOUND_PRIME=m > CONFIG_SOUND_MSNDCLAS=m > CONFIG_MSNDCLAS_INIT_FILE="/etc/sound/msndinit.bin" > CONFIG_MSNDCLAS_PERM_FILE="/etc/sound/msndperm.bin" > CONFIG_SOUND_MSNDPIN=m > CONFIG_MSNDPIN_INIT_FILE="/etc/sound/pndspini.bin" > CONFIG_MSNDPIN_PERM_FILE="/etc/sound/pndsperm.bin" > CONFIG_SOUND_OSS=m > # CONFIG_SOUND_TRACEINIT is not set > CONFIG_SOUND_DMAP=y > CONFIG_SOUND_SSCAPE=m > CONFIG_SOUND_VMIDI=m > CONFIG_SOUND_TRIX=m > CONFIG_SOUND_MSS=m > CONFIG_SOUND_MPU401=m > CONFIG_SOUND_PAS=m > CONFIG_SOUND_PSS=m > CONFIG_PSS_MIXER=y > # CONFIG_PSS_HAVE_BOOT is not set > CONFIG_SOUND_SB=m > CONFIG_SOUND_YM3812=m > CONFIG_SOUND_UART6850=m > CONFIG_SOUND_AEDSP16=m > CONFIG_SC6600=y > CONFIG_SC6600_JOY=y > CONFIG_SC6600_CDROM=4 > CONFIG_SC6600_CDROMBASE=0 > CONFIG_SOUND_KAHLUA=m > CONFIG_AC97_BUS=m > CONFIG_HID_SUPPORT=y > CONFIG_HID=m > # CONFIG_HID_DEBUG is not set > CONFIG_HIDRAW=y > > On Wed, Dec 17, 2008 at 9:40 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 17 Dec 2008 08:37:58 +0100, > Klaus Schulz wrote: > > > > Hi there. > > > > Can anybody tell me if /proc/asound/card0/stream0 is not existing any > longer > > and what has replaced it? > > It's built only when kernel config CONFIG_SND_VERBOSE_PROCFS=y. > In the case of alsa-driver tarball, this is enabled unless you add > --disable-verbose-procfs configure option. > > Takashi > >

1 0

[alsa-devel] Sigmatel patch for missing
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi folks. Below patch seems to be missing in 1.0.18a! Note that also MCP51 seemed to work afterwards. It needs to be ported to 1.0.18.a though. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3036 Can anybody let me know how the patch would look like in latest driver rev. THX Klaus --- patch_sigmatel.c.orig 2007-09-15 17:57:07.000000000 -0700 +++ patch_sigmatel.c 2007-09-15 17:50:26.000000000 -0700 @@ -2304,6 +2304,15 @@ stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + switch (codec->subsystem_id) { + case 0x107b0317: // Gateway MX3412, MT3423 + case 0x107b0318: // Gateway MT3707, ML3109 (ATI SB450) + case 0x10de026c: // Gateway MX3410 + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_POWER_STATE, 0x00); + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); + break; + } + if (spec->gpio_mute) { stac922x_gpio_mute(codec, 0, 0); stac922x_gpio_mute(codec, 1, 0);

2 1

[alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi there. Can anybody tell me if /proc/asound/card0/stream0 is not existing any longer and what has replaced it? THX a lot Klaus

2 1

Re: [alsa-devel] Trying to figure out what went wrong with my HDMI.
by Clemens Ladisch 17 Dec '08

17 Dec '08

Mark Jenks wrote: > On Tue, Dec 16, 2008 at 7:24 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > Use "plughw:0,3" instead of "hw:0,3". > > What is the difference? Plughw automatically converts to a sample format that the device supports. > And why did hw:0,3 work before? Probably your old hardware did support mono playback. > If I want to use it in Myth, would it be ALSA:plughw:0,3 then? Yes. (It is possible that Myth can do resampling and channel rerouting by itself; in this case, hw would also work.) HTH Clemens

1 0

[alsa-devel] alsa-utils: alsactl restore not working after system restart
by Klaus Schulz 16 Dec '08

16 Dec '08

Hi there. I am running Ubuntu 8.10 - and latest ALSA. After restart amixer defaults to default settings. I am not really sure if this is Ubuntu/Debian specific or an Alsa issue. It seems that either a link /var/lib/alsa/asound.state to /etc/asound.state is needed or that /etc/init.d/alsa-utils needs to be updated within the restore section accordingly. With the link workaround I managed to get it working. Cheers Klaus

3 2

[alsa-devel] [PATCH] 1/2 ALSA: ca0106 support 44100Hz playback to spdif
by Ben Stanley 16 Dec '08

16 Dec '08

>From 241939bdf80de14e204d6fcb43d850d33bdeda20 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Tue, 16 Dec 2008 23:47:36 +1100 Subject: [PATCH] ALSA: ca0106 support 44100Hz playback to spdif This patch provides support for playback of 44100Hz sampled material in 16 and 32 bit sample depths through the digital spdif/iec958 connection/s available on the ca0106 card. I have re-worked the patch to address comments from a previous review by Takashi [1]. In particular, I have incorporated spin-locking to address a race condition that Takashi was concerned about. This code has been running on my MythTV box for the last fortnight without apparent ills. I have re-based the code a bit for submission. This patch is against alsa-kmirror. [1] http://article.gmane.org/gmane.linux.alsa.devel/55825/match=takashi+ca0106+… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 13 ++- pci/ca0106/ca0106_main.c | 402 ++++++++++++++++++++++++++++++++++++++------- 2 files changed, 351 insertions(+), 64 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 3faccb6..2ffec74 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.22 + * Version: 0.0.23 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -49,6 +49,8 @@ * Implement support for Line-in capture on SB Live 24bit. * 0.0.22 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.23 + * Add support for playback sampling rate and format constraints. * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -644,6 +646,8 @@ #include "ca_midi.h" +#define DRVNAME "snd-ca0106" + struct snd_ca0106; struct snd_ca0106_channel { @@ -659,6 +663,7 @@ struct snd_ca0106_pcm { struct snd_pcm_substream *substream; int channel_id; unsigned short running; + unsigned short hw_reserved; }; struct snd_ca0106_details { @@ -688,6 +693,7 @@ struct snd_ca0106 { unsigned short model; /* subsystem id */ spinlock_t emu_lock; + spinlock_t pcm_lock; struct snd_ac97 *ac97; struct snd_pcm *pcm; @@ -707,6 +713,11 @@ struct snd_ca0106 { struct snd_ca_midi midi2; u16 spi_dac_reg[16]; + + unsigned char count_pb_44100_chan; + unsigned char count_pb_non_44100_chan; + unsigned char count_pb_S16_chan; + unsigned char count_pb_S32_chan; }; int snd_ca0106_mixer(struct snd_ca0106 *emu); diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 6ac1936..bc78312 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton <James(a)superbug.demon.co.uk> * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.25 + * Version: 0.0.26 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -83,9 +83,14 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * 0.0.25 * Powerdown SPI DAC channels when not in use + * 0.0.26 Ben Stanley + * Added support for output at 44100Hz rate (SPDIF only). + * Implemented constraints system for output rate and format. * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. + * Some programs fail to produce sound output (tested on SPDIF). See + * http://thread.gmane.org/gmane.linux.alsa.devel/55384/focus=55410 * -- * * TODO: @@ -145,6 +150,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/ac97_codec.h> #include <sound/info.h> @@ -285,9 +291,9 @@ static struct snd_pcm_hardware snd_ca0106_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000), - .rate_min = 48000, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, .rate_max = 192000, .channels_min = 2, //1, .channels_max = 2, //6, @@ -319,6 +325,110 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .fifo_size = 0, }; +static unsigned int all_spdif_playback_rates[] = + {44100, 48000, 96000, 192000}; + +static int hw_rule_playback_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + int mask; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->spdif_enable) { + /* Compute the mask applied to all_spdif_playback_rates */ + if (chip->count_pb_44100_chan) + mask = 0x1; + else if (chip->count_pb_non_44100_chan) + mask = 0xE; + else + mask = 0xF; + } else { + /* 44100Hz is not supported for DAC. + The DAC cannot be clocked at 44100, only 48000 and 96000 + from the CA0106 chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + mask = 0xE; + } + snd_printdd("snd_hw_rule_playback_rate: any_44100=%d, " + "any_non_44100=%d, mask=0x%X, spdif=%d\n", + chip->count_pb_44100_chan, + chip->count_pb_non_44100_chan, + mask, chip->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_interval_list(hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE), + ARRAY_SIZE(all_spdif_playback_rates), + all_spdif_playback_rates, mask); +} + +static int hw_rule_playback_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_ca0106 *chip = rule->private; + struct snd_mask fmt; + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + int result; + unsigned long flags; + if (snd_BUG_ON(!chip)) + return -EINVAL; + snd_mask_none(&fmt); + + spin_lock_irqsave(&chip->pcm_lock, flags); + if (chip->count_pb_S16_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + else if (chip->count_pb_S32_chan) + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + else { + /* No format yet chosen, so both formats are available. */ + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set(&fmt, SNDRV_PCM_FORMAT_S32_LE); + } + result = snd_mask_refine(f, &fmt); + snd_printdd("snd_hw_rule_playback_format: any_S16=%d, any_S32=%d, " + "refined_fmt=0x%X, avail_fmt=0x%X, changed=%d\n", + chip->count_pb_S16_chan, + chip->count_pb_S32_chan, f->bits[0], fmt.bits[0], + result); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return result; +} + +void snd_ca0106_rebuild_playback_channel_counters(struct snd_ca0106 *chip, + int skip_channel) +{ + /* Re-build the counters of rate=44100 and the formats + for use in constraints processing. Does not process the + current_channel (this must be dealt with in the calling context). + To process all channels, set skip_channel=-1. + This must be called with chip->pcm_lock held. */ + + int chi; + struct snd_ca0106_channel *pchannel; + struct snd_pcm_runtime *runtimei; + + chip->count_pb_44100_chan = chip->count_pb_non_44100_chan = 0; + chip->count_pb_S16_chan = chip->count_pb_S32_chan = 0; + for (chi = 0; chi < 4; ++chi) { + if (chi == skip_channel) + continue; + pchannel = &(chip->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm + || !pchannel->epcm->hw_reserved) + continue; + runtimei = pchannel->epcm->substream->runtime; + chip->count_pb_44100_chan += runtimei->rate == 44100; + chip->count_pb_non_44100_chan += runtimei->rate != 44100; + chip->count_pb_S16_chan += + runtimei->format == SNDRV_PCM_FORMAT_S16_LE; + chip->count_pb_S32_chan += + runtimei->format == SNDRV_PCM_FORMAT_S32_LE; + } +} + unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, unsigned int reg, unsigned int chn) @@ -393,7 +503,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, int status; int retry; if ((reg > 0x7f) || (value > 0x1ff)) { - snd_printk(KERN_ERR "i2c_write: invalid values.\n"); + snd_printk(KERN_ERR DRVNAME "i2c_write: invalid values.\n"); return -EINVAL; } @@ -463,7 +573,20 @@ static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) { + struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106 *chip = epcm->emu; + /* FIXME how to tell which case to use? */ + /* struct snd_ca0106_channel *channel = + &(chip->playback_channels[epcm->channel_id]); */ + /* struct snd_ca0106_channel *channel = + &(chip->capture_channels[epcm->channel_id]); */ + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); kfree(runtime->private_data); + runtime->private_data = 0; + /* *channel = 0; */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); } static const int spi_dacd_reg[] = { @@ -487,19 +610,21 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) return -ENOMEM; + spin_lock_irqsave(&chip->pcm_lock, flags); epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_playback_hw; channel->emu = chip; @@ -509,10 +634,24 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + goto error; + err = snd_pcm_hw_constraint_step(runtime, + 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_playback_rate, (void *)chip, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto error; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_playback_format, (void *)chip, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + goto error; snd_pcm_set_sync(substream); if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { @@ -522,9 +661,12 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); if (err < 0) - return err; + goto error; } - return 0; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -532,8 +674,12 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + spin_unlock_irqrestore(&chip->pcm_lock, flags); if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { const int reg = spi_dacd_reg[epcm->channel_id]; @@ -574,6 +720,7 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]); struct snd_ca0106_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); @@ -584,10 +731,11 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre epcm->emu = chip; epcm->substream = substream; epcm->channel_id=channel_id; - + + spin_lock_irqsave(&chip->pcm_lock, flags); runtime->private_data = epcm; runtime->private_free = snd_ca0106_pcm_free_substream; - + runtime->hw = snd_ca0106_capture_hw; channel->emu = chip; @@ -596,13 +744,16 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->use = 1; //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); //channel->interrupt = snd_ca0106_pcm_channel_interrupt; - channel->epcm = epcm; + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; + goto error; //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) - return err; - return 0; + goto error; + err = 0; +error: + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return err; } /* close callback */ @@ -610,9 +761,13 @@ static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream) { struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + spin_lock_irqsave(&chip->pcm_lock, flags); + epcm = runtime->private_data; chip->capture_channels[epcm->channel_id].use = 0; /* FIXME: maybe zero others */ + spin_unlock_irqrestore(&chip->pcm_lock, flags); return 0; } @@ -647,6 +802,24 @@ static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream /* hw_free callback */ static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) { + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime; + struct snd_ca0106_pcm *epcm; + unsigned long flags; + + spin_lock_irqsave(&chip->pcm_lock, flags); + runtime = substream->runtime; + epcm = runtime->private_data; + epcm->hw_reserved = 0; + + snd_ca0106_rebuild_playback_channel_counters(chip, epcm->channel_id); + + snd_printdd(KERN_DEBUG DRVNAME ": free_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + chip->count_pb_44100_chan, chip->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&chip->pcm_lock, flags); + return snd_pcm_lib_free_pages(substream); } @@ -668,53 +841,149 @@ static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) { struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_ca0106_pcm *epcm = runtime->private_data; - int channel = epcm->channel_id; - u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + struct snd_pcm_runtime *runtime = substream->runtime, *runtimei; + struct snd_ca0106_pcm *epcm; + struct snd_ca0106_channel *pchannel; + int channel, chi; + unsigned char count_pb_44100_chan, count_pb_non_44100_chan; + unsigned char count_pb_S16_chan, count_pb_S32_chan; + u32 *table_base; u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; u32 hcfg_set = 0x00000000; u32 hcfg; - u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_mask = 0xFF0000; u32 reg40_set = 0; u32 reg40; - /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ - u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_mask; + u32 reg71_shift; u32 reg71_set = 0; u32 reg71; int i; + unsigned long flags; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* FIXME CLEAN UP IF spdif_enable IS CHANGED WHILE CHANNELS ARE OPENED + * OR PREVENT THIS FROM HAPPENING. */ + if (emu->spdif_enable) + reg71_shift = 24; /* SPDIF Output Rate */ + else + reg71_shift = 16; /* I2S Output Rate */ + reg71_mask = 0x3 << reg71_shift; + + /*printk(KERN_DEBUG DRVNAME ": prepare_playback: " + "channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld,period_size=%ld, periods=%u, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1));*/ + /*printk("dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr,runtime->dma_area, table_base);*/ + /*printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr,emu->buffer.area, emu->buffer.bytes);*/ + /* We are forced to build the settings for all the channels. */ + spin_lock_irqsave(&emu->pcm_lock, flags); + epcm = runtime->private_data; + channel = epcm->channel_id; + + snd_ca0106_rebuild_playback_channel_counters(emu, channel); + + /* Determine the effects of the new settings for validation. + Note that nothing is recorded permanently + until the new settings are validated. */ + count_pb_44100_chan = + emu->count_pb_44100_chan + runtime->rate == 44100; + count_pb_non_44100_chan = + emu->count_pb_non_44100_chan + runtime->rate != 44100; + count_pb_S16_chan = emu->count_pb_S16_chan + + runtime->format == SNDRV_PCM_FORMAT_S16_LE; + count_pb_S32_chan = emu->count_pb_S32_chan + + runtime->format == SNDRV_PCM_FORMAT_S32_LE; + /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ - switch (runtime->rate) { - case 44100: - reg40_set = 0x10000 << (channel<<1); - reg71_set = 0x01010000; - break; - case 48000: - reg40_set = 0; - reg71_set = 0; - break; - case 96000: - reg40_set = 0x20000 << (channel<<1); - reg71_set = 0x02020000; - break; - case 192000: - reg40_set = 0x30000 << (channel<<1); - reg71_set = 0x03030000; - break; - default: - reg40_set = 0; - reg71_set = 0; - break; + for (chi = 0; chi < 4; ++chi) { + pchannel = &(emu->playback_channels[chi]); + if (!pchannel->use || !pchannel->epcm) + continue; + if (!pchannel->epcm->hw_reserved && chi != channel) + continue; + runtimei = pchannel->epcm->substream->runtime; + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtimei->rate) { + case 44100: + /* 44100Hz is not supported for DAC. The DAC cannot be + clocked at 44100, only 48000 and 96000 from the CA0106 + chip. The SPDIF can be clocked at 44100. + It is a hardware limitation. */ + if (emu->spdif_enable) { + /* When using 44100, *all* channels + must be set to that rate. */ + reg40_set |= 0x550000; + reg71_set |= 0x1 << reg71_shift; + break; + } else { + printk(KERN_ERR DRVNAME + "prepare_playback: " + "44100Hz is invalid for DAC.\n"); + } + case 48000: + /* reg40_set &= !(0x1 << (chi<<1)); */ + /* reg71_set &= !(0x1 << reg71_shift); */ + break; + case 96000: + reg40_set |= 0x20000 << (chi<<1); + reg71_set |= 0x2 << reg71_shift; + break; + case 192000: + reg40_set |= 0x30000 << (chi<<1); + reg71_set |= 0x3 << reg71_shift; + break; + default: + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + ": prepare_playback: " + "Bad sampling frequency %dHz.\n", + runtime->rate); + return -EINVAL; + } } - /* Format is a global setting */ - /* FIXME: Only let the first channel accessed set this. */ + if (count_pb_44100_chan && count_pb_non_44100_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sampling rate %dHz" + " conflicts with other selected sampling rates.\n", + runtime->rate); + return -EINVAL; + } + if (count_pb_S16_chan && count_pb_S32_chan) { + spin_unlock_irqrestore(&emu->pcm_lock, flags); + printk(KERN_ERR DRVNAME + "prepare_playback: requested sample format %d" + " conflicts with other selected sample formats.\n", + runtime->format); + return -EINVAL; + } + /* Requested sample rate and format are now validated. + Commit the change. */ + emu->count_pb_44100_chan = count_pb_44100_chan; + emu->count_pb_non_44100_chan = count_pb_non_44100_chan; + emu->count_pb_S16_chan = count_pb_S16_chan; + emu->count_pb_S32_chan = count_pb_S32_chan; + epcm->hw_reserved = 1; + + snd_printdd(KERN_DEBUG DRVNAME ": prepare_playback: any_44100=%d, " + "any_non_44100=%d, spdif=%d.\n", + emu->count_pb_44100_chan, emu->count_pb_non_44100_chan, + emu->spdif_enable); + spin_unlock_irqrestore(&emu->pcm_lock, flags); + + /* Format is a global setting. */ + /* Only the first channel accessed can set this + (enforced by constraints). */ switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: hcfg_set = 0; @@ -736,10 +1005,13 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) reg71 = (reg71 & ~reg71_mask) | reg71_set; snd_ca0106_ptr_write(emu, 0x71, 0, reg71); - /* FIXME: Check emu->buffer.size before actually writing to it. */ - for(i=0; i < runtime->periods; i++) { - table_base[i*2] = runtime->dma_addr + (i * period_size_bytes); - table_base[i*2+1] = period_size_bytes << 16; + table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + if (!snd_BUG_ON(8*16*channel+runtime->periods*2 > emu->buffer.bytes)) { + for (i = 0; i < runtime->periods; i++) { + table_base[i*2] = runtime->dma_addr + + (i * period_size_bytes); + table_base[i*2+1] = period_size_bytes << 16; + } } snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); @@ -1335,6 +1607,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, chip->irq = -1; spin_lock_init(&chip->emu_lock); + spin_lock_init(&chip->pcm_lock); chip->port = pci_resource_start(pci, 0); if ((chip->res_port = request_region(chip->port, 0x20, @@ -1351,8 +1624,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - - /* This stores the periods table. */ + + /* This stores the periods table. FIXME Only the first 512 bytes + appear to be used (for playback buffers) */ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { snd_ca0106_free(chip); return -ENOMEM; @@ -1363,8 +1637,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 - printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - pci->revision, chip->serial); + printk(KERN_INFO DRVNAME ": Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); @@ -1378,7 +1652,9 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } chip->details = c; if (subsystem[dev]) { - printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + printk(KERN_INFO DRVNAME + ": Sound card name=%s, subsystem=0x%x. " + "Forced to subsystem=0x%x\n", c->name, chip->serial, subsystem[dev]); } -- 1.5.4.3

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[alsa-devel] [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1
by Ben Stanley 16 Dec '08

16 Dec '08

>From fe2e6692ea6fcdf93d724b8aa48e7433994e011c Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Mon, 15 Dec 2008 23:38:12 +1100 Subject: [PATCH] ALSA: ca0106 MSI K8N Diamond MB spi_dac 2->1 This patch removes an inconsistency that became apparent when I documented the fields of snd_ca0106_details. spi_dac is always used in a 'boolean' sense, so this cleanup should make no difference. Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106_main.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/pci/ca0106/ca0106_main.c b/pci/ca0106/ca0106_main.c index 88fbf28..6ac1936 100644 --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 2 } , + .spi_dac = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- 1.5.4.3

2 1

[alsa-devel] [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct
by Ben Stanley 16 Dec '08

16 Dec '08

>From dada1c7fe866b860f96d56315648cd5704675aa1 Mon Sep 17 00:00:00 2001 From: Ben Stanley <Ben.Stanley(a)exemail.com.au> Date: Fri, 12 Dec 2008 09:47:13 +1100 Subject: [PATCH] ALSA: ca0106 Add comments to snd_ca0106_details struct Takashi wrote an email [1] explaining the fields of snd_ca0106_details, so I captured the information into the ca0106.h header file. [1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_ty… Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- pci/ca0106/ca0106.h | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/pci/ca0106/ca0106.h b/pci/ca0106/ca0106.h index 74175fc..3faccb6 100644 --- a/pci/ca0106/ca0106.h +++ b/pci/ca0106/ca0106.h @@ -664,10 +664,14 @@ struct snd_ca0106_pcm { struct snd_ca0106_details { u32 serial; char * name; - int ac97; - int gpio_type; - int i2c_adc; - int spi_dac; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + int spi_dac; /* spi_dac=1 adds the mute switch for each analog + output, front, rear, etc. */ }; // definition of the chip-specific record -- 1.5.4.3

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[alsa-devel] Trying to figure out what went wrong with my HDMI.
by Mark Jenks 16 Dec '08

16 Dec '08

I have a M3N78 Pro motherboard, and am running alsa 1.0.18a on it. I had HDMI audio working on it, but for some reason it has stopped. I really didn't change too much about the computer, other then take out a hvr-1800, and put in a hvr-1600. I did try to go up to .27 kernel, but had other problems and went back to .26. Recompiled alsa 1.0.18a and even tried multiple versions of the nvidia drivers. (173, 177).. And now I have just tried the snapshot with the same results. I believe it was recognised as a ALC883 before, and not is is a ALC1200, but I'm not completely sure about it. Help! :) -Mark aplay -l shows: card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So, everything looks good there. If I try and do an aplay, I get. #aplay -D hw:0,3 /usr/share/sounds/alsa/Noise.wav Playing WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:966: Channels count non available I noticed that I am missing hdmi in my aplay -L. When it worked, it was there. # aplay -L front:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default

2 1

[alsa-devel] ASoC: help needed for capture DAPM (twl4030)
by Peter Ujfalusi 16 Dec '08

16 Dec '08

Hello, I need some help/advice on how to implement the DAPM routings for twl4030 capture path. There are at least two issues which I'm wondering at the moment. [1] Mux for the analog paths As with other parts, this is implemented as a bitfield, where only one bit can be enabled at the time (0x0 being no routed inputs). ADCL <-|- Main Mix (0x1) |- Headset mic (0x2) |- AUXL, LineL (0x4) |- Carkit mic (0x8) Invalid values: 0x3, 0x5, 0x6, 0x7 ADCR <-|- Sub mic (0x1) |- AUXR, LineR (0x4) Invalid values: 0x2, 0x3 Since the playback path also has similar 'mux as not enum but bitfield' types, I'm thinking of creating a new type for soc based on the soc_enum. This new type would store the valid enum texts and the corresponding value in the register (with mask). This could be used in very flexible way. It can be also be used instead of the normal enum types (altrough it does not make sense). But this can wait for now. [2] ADC versus digital microphone TWL has two ADCs (ADCL, ADCR), the routing looks something like this, when the ADCs are in use: InputL mux -> AmpliL -> ADCL |-> ATXL1_PGA -> Mic channel1 (I2S, TDM) |-> ATXL2_PGA -> Mic channel2 (TDM) InputR mux -> AmpliR -> ADCR |-> ATXR1_PGA -> Mic channel3 (I2S, TDM) |-> ATXR2_PGA -> Mic channel4 (TDM) When the digital microphones are in use: Digimic0 Left -> ATXL1_PGA -> Mic channel1 (I2S, TDM) Digimic0 Right -> ATXR1_PGA -> Mic channel3 (I2S, TDM) Digimic1 Left -> ATXL2_PGA -> Mic channel2 (TDM) Digimic1 Right -> ATXR2_PGA -> Mic channel4 (TDM) There is a single point of selection for Mic channel1,3 and for channel2,4. So ATXL1_PGA and ATXR1_PGA receives the data either from the ADCs or from the digimic0, ATXL2_PGA and ATXR2_PGA receives the data either from the ADCs or from the digimic1. As you can see, if the digital mic is selected the ADC can be powered off, since they are not needed. How to map this with DAPM? Is it mandatory to have the SND_SOC_DAPM_INPUT(digimic) to be connected to SND_SOC_DAPM_ADC? I have been thinking to create four 'ghost ADC' for the four channel. Than when building the route the real ADCs can receive switches to handle the power management. Thank you, -- Péter

2 1

[alsa-devel] [PATCH] ALSA: hda - Add automatic model setting for Samsung Q45
by Luke Yelavich 16 Dec '08

16 Dec '08

Have the Samsung Q45 (144d:c510) select ALC262_HIPPO by default Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4afa0c9..3b2a793 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10541,6 +10541,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), -- 1.5.6.3

2 1

[alsa-devel] HDMI problem, more info.
by Mark Jenks 16 Dec '08

16 Dec '08

More info about the M3n78 Pro motherboard that I am having problems with it showing properly in 'aplay -L'. I turned on ./configure -with-debug=verbose ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x6 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_intel.c:1469: azx_pcm_prepare: bufsize=0x4400, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:874: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x4011 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x6 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x2 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x3 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x4 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x5 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 ALSA /data2/installs/alsa/temp/alsa-driver/pci/hda/hda_codec.c:887: hda_codec_cleanup_stream: NID=0x8 # cat /proc/asound/card0/codec#3 Codec: Generic 10de NVIDIA MCP78 HDMI Address: 3 Vendor Id: 0x10de0002 Subsystem Id: 0x10de0101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x18560110: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560121: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560122: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560123: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560124: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c

1 0

[alsa-devel] [PATCH] ALSA: Fix declaration of sound_class
by Mark Brown 15 Dec '08

15 Dec '08

Include sound/core.h in sound_core.c so that sound_class is declared before it is defined, avoiding it looking like it should be static. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/sound_core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 10ba421..2b302bb 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/err.h> +#include <sound/core.h> #ifdef CONFIG_SOUND_OSS_CORE static int __init init_oss_soundcore(void); -- 1.5.6.5

2 2

[alsa-devel] Xrun stack trace for 1010LT
by Pete cat 15 Dec '08

15 Dec '08

Im trying to get xrun free recording from a pair of 1010LTs (ice1712) and ive resorted to setting xrun_debug to record stack traces on xruns. Im getting xruns every few minutes (between 2 and 15 minute gaps between single xrun events) and every xrun produces identical stack traces like this: [17127.486287] Pid: 775, comm: IRQ-3 Not tainted 2.6.24.1 #2 [17127.486309] [<c02ade37>] snd_pcm_period_elapsed+0x247/0x2f0 [17127.486325] [<c010312e>] __switch_to+0x9e/0x150 [17127.486336] [<c0128674>] finish_task_switch+0x34/0xd0 [17127.486350] [<c02c2119>] snd_ice1712_interrupt+0xa9/0x1a0 [17127.486364] [<c016ab7c>] handle_IRQ_event+0x5c/0x100 [17127.486379] [<c016b1ba>] thread_simple_irq+0x4a/0x90 [17127.486390] [<c016bb09>] do_irqd+0x229/0x290 [17127.486404] [<c016b8e0>] do_irqd+0x0/0x290 [17127.486411] [<c0141ef2>] kthread+0x42/0x70 [17127.486418] [<c0141eb0>] kthread+0x0/0x70 [17127.486425] [<c0105857>] kernel_thread_helper+0x7/0x10 [17127.486437] ======================= Running the rt kernel (Linux ubuntu-studio 2.6.24.1 #2 SMP PREEMPT RT Thu Dec 4 21:30:26 GMT 2008 i686 GNU/Linux) - although the original installation was Ubuntu studio, the kernel is recompiled. The xruns seem to be all on the same card. IRQs are set to priority of 99 with chrt. Software for record is arecord, modified to use a 10 second ring buffer which is written to disk in a separate thread. I was hoping that from the stack trace I could work out what was causing the xruns, but from the above it dosent look like there is anything causing them. However this is the first time I am using stack traces so maybe im missing something. Can anyone point to the fault for me? Thanks, Pete

4 11

[alsa-devel] Intel 82801I (ICH9 Family) HD Audio Controller on a HP dv4 computer
by Jean-Pierre ANDRE 15 Dec '08

15 Dec '08

Hi, I am getting difficulty setting up ALSA for my new HP portable computer. lspci displays the audio chip as : Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) and /proc/asound displays the codec as IDT 92HD71B7X I have uploaded the detailed hardware and configuration parameters to http://pagesperso-orange.fr/b.andre/visits.html/92HD71B7X I am using Fedora 10 x86_64 without pulseaudio and with the latest alsa-driver : alsa-driver-1.0.18a.16.g4012f.139.g6e583.tar.bz2 I have googled for a similar problem, but did not get to an actual fix, the following thread appears to deal with a most similar situation : http://forums.opensuse.org/pre-release-beta/399731-beta-5-no-sound.html After several tries, I have set modprobe.conf as alias snd-card-0 snd-hda-intel options sound slots=snd-hda-intel options snd-hda-intel model=hp-m4 single_cmd=1 enable_msi=1 Results : aplay freezes, with some sound output which is the continual repeat of the initial expected sound (some fraction of a second repeated). However if I boot with pci=noacpi the output is correct (I do not understand the relation of power management to sound, there must be one as I see : HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 when I unload/reload alsa) WITHOUT the pci=noacpi option : - mplayer outputs the sound correctly (does mplayer bypass alsa ?) - xine outputs a chopped sound (some 100ms OK, some 100ms silent, some 100ms OK, etc, with a period which could be 200..500ms) - most players (eg Real player) just freeze. I would say that when some sound buffer gets empty, the driver does not get the information, so it does not fill the buffer again. I have done some digging into the code. I see that this 92HD71B7X chip is supported for HP dv5 and dv7 computers (subsystem ID 103c30f2 and 103C30f4), but mine is a HP dv4 showing a subsystem ID 103c30f7, so it must be somewhat different and not supported yet. Can anybody suggest some fix ? If needed I can apply patches to the code and do some testing so that my configuration gets supported in subsequent versions. Regards Jean-Pierre

2 6

Re: [alsa-devel] alsa 1.0.18 does not work for Intel onboard cards
by Takashi Iwai 15 Dec '08

15 Dec '08

At Mon, 15 Dec 2008 07:55:22 -0200, Paulo Cavalcanti wrote: > > On Mon, Dec 15, 2008 at 5:31 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Sun, 14 Dec 2008 09:29:03 -0200, > Paulo Cavalcanti wrote: > > > > On Sun, Dec 14, 2008 at 7:12 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Sat, 13 Dec 2008 13:52:24 -0200, > > Paulo Cavalcanti wrote: > > > > > > On Sat, Dec 13, 2008 at 1:23 PM, Paulo Cavalcanti < > promac(a)gmail.com> > > wrote: > > > > > > > Hi, > > > > > > > > alsa driver 1.0.18 has been a nightmare for me. > > > > > > > > I had a perfectly working sound for my Intel D975XBX2 when using > > > > alsa driver 1.017. Going to 1.0.18 just muted my front sound. > > > > Well, maybe not everybody needs a front sound output, but I just > > > > stick with the 1.01.17 driver. > > > > > > > > > > > Then, I tried an Intel DG45ID mobo, which does not have > > > any sound at all, with the later alsa driver 1.0.18a (detected as > a > > Generic > > > DG45). > > > > > > Going back to alsa driver 1.0.17, from the stock Fedora 10 kernel > > > (my card is then detected as an IDT 92HD73E1X5), > > > I have an analogic sound working, but Master, Headphones, and PCM > > > channels have to be above 90%, for anyone being able to hear > anything. > > > Therefore, the volume is compressed in the upper 10% of the scale. > > > > > > Furthermore, I have no mic, and no input working (I would like to > feed > > the > > > output > > > of my capture card to the input source of the sound card) . > > > It seems to me that the input (blue jack) is configured as > output, > > > in the 1.0.17 alsa driver. > > > > > > I do not know what else I can try, maybe some alsa snapshot, > > > to fix some of these issues. > > > > Try the snapshot on > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > alsa-driver-snapshot.tar.gz > > > > Thanks, Takashi > > > > but it did not change anything for the Intel DG45ID. > > This is a Genuine Intel mother board. > > > > The sound card is detected as "Generic 8086 Intel G45 Develk", > > but I do not hear anything, even with all mixer channel at the maximum > > volumes. > > > > This is a Fedora 10, with the latest alsa packages, and > > everything seems OK, the applications work, but I do no hear anything. > > > > With the alsa driver from the latest Fedora kernel > > > > Linux version 2.6.27.7-134.fc10.x86_64 > > > > the card is detected as IDT 92HD73E1X5 and, > > > > although I have to put all the mixer channels at the maximum, the sound > > output works fine, even mid emulation. > > > > During the snapshot compilation, I got dozens of this warning: > > > > > ---------------------------------------------------------------------------------- > > /home/roma/redhat/BUILD/alsa-driver/include/config1.h:65:1: warning: > > "CONFIG_SND_AC97_POWER_SAVE_DEFAULT" redefined > > In file included from <command-line>:0: > > ./include/linux/autoconf.h:260:1: warning: this is the location of the > > previous definition > > CC [M] /home/roma/redhat/BUILD/alsa-driver/misc/ac97_bus.o > > In file included from /home/roma/redhat/BUILD/alsa-driver/include/ > config.h:6, > > from /home/roma/redhat/BUILD/alsa-driver/include/ > > adriver.h:25, > > from /home/roma/redhat/BUILD/alsa-driver/misc/ > ac97_bus.c:1: > > > ----------------------------------------------------------------------------------- > > It's no big issue, I guess. > > > This is the output of alsa-info, with the stock driver (1.0.17): > > Run alsa-info with --no-upload option on both working (1.0.17) and > non-working versions, and attach the generated files. > > This is for the Intel D975XBX2, which lost the front sound with 1.0.18: > Soundcard is detected as Sigmatel STAC9274D. Which I/O are you referring to? The headphone output seems unplugged, and thus the outputs are only from other line-outs, as far as I see the codec info. Takashi

1 0

[alsa-devel] : Can't compile alsa-driver-1.0.18a on x86_64 arch
by Mysth-R 15 Dec '08

15 Dec '08

Hi, I am trying to compile the lastest alsa-driver, but I got an error : patching file usbusx2y.c copying file alsa-kernel/usb/usx2y/usbusx2yaudio.c patching file usbusx2yaudio.c Hunk #11 succeeded at 1057 (offset 1 line). copying file alsa-kernel/usb/usx2y/usx2yhwdeppcm.c patching file usx2yhwdeppcm.c make[3]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a/usb/usx2y » make[2]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a/usb » make[1]: quittant le répertoire « /home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a » make -C /lib/modules/2.6.24.7-rt7/build SUBDIRS=/home/mysthr2/Logiciels/musique/alsa-driver-1.0.18a CPP="gcc -E" CC="gcc" modules make[1]: entrant dans le répertoire « /usr/src/linux-2.6.24.7-rt7 » Makefile:510: /usr/src/linux-2.6.24.7-rt7/arch/x86/Makefile: Aucun fichier ou dossier de ce type make[1]: *** Pas de règle pour fabriquer la cible « /usr/src/linux-2.6.24.7-rt7/arch/x86/Makefile ». Arrêt. make[1]: quittant le répertoire « /usr/src/linux-2.6.24.7-rt7 » make: *** [compile] Erreur 2 Here is my ./configure : ./configure --prefix=/usr --with-cards=hda-intel usb-caiaq usb-audio hrtimer rtctimer hpet --with-sequencer=yes --with-card-options=all --with-oss=no--with-kernel=/usr/src/linux-2.6.24.7-rt7/ CFLAGS="-march=nocona -pipe -O2" Thank you for your help ! cheers, Mysth-R -- * *************************************************************************************** * {^_^} Mysth-R {^_^} * <= Aide Auditive => * * http://myspace.com/mysthr * http://myspace.com/aideauditive * http://mysthr.free.fr/Joomla => Site dédié à l'audio sous Fedora/PlanetCCRMA. * ***************************************************************************************

1 0

[alsa-devel] Latest snapshot - reliable source?
by Klaus Schulz 15 Dec '08

15 Dec '08

Hello Takashi. Can I regard your ftp link below as a kind of reliable source where to get the latest snapshot at any time? Can I also expect that it is compatible to latest official driver release? Because than I'd ask people in (worst) case of trouble to give it a try before spamming the mailing list. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… THX \Klaus

2 3

[alsa-devel] alsa 1.0.18 does not work for Intel onboard cards
by Paulo Cavalcanti 15 Dec '08

15 Dec '08

Hi, alsa driver 1.0.18 has been a nightmare for me. I had a perfectly working sound for my Intel D975XBX2 when using alsa driver 1.017. Going to 1.0.18 just muted my front sound. Well, maybe not everybody needs a front sound output, but I just stick with the 1.01.17 driver. Then, I -- Paulo Roma Cavalcanti LCG - UFRJ

2 4

[alsa-devel] [PATCH] SOC/davinci: switch DPRINTK to pr_debug()
by Alexander Beregalov 14 Dec '08

14 Dec '08

Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/soc/davinci/davinci-pcm.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bc83e1c..74abc9b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> @@ -24,13 +25,6 @@ #include "davinci-pcm.h" -#define DAVINCI_PCM_DEBUG 0 -#if DAVINCI_PCM_DEBUG -#define DPRINTK(x...) printk(KERN_DEBUG x) -#else -#define DPRINTK(x...) -#endif - static struct snd_pcm_hardware davinci_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; - DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " - "period_size=%x\n", lch, dma_pos, period_size); + pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " + "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; @@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_writecombine(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, (void *) buf->addr, size); + pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " + "size=%d\n", (void *) buf->area, (void *) buf->addr, size); if (!buf->area) return -ENOMEM;

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[alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

I have just bought an Aspire one A150, XP version, as it was the only available here, and installed ubuntu on it. Bugs I discovered so far: ** 1 - embedded controler works in polling mode, due to this: [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. Maybe this is the reason for the fact that gnome power manager freezes when I unplug the AC, and freezes often when I try to see battery status. (Note: same is seen on my acer aspire 5720) ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... wireless more or less works, but kernel log is full of backtraces. Was able to connect to my WPA2 access point. Sometimes wireless fails completely, especially after suspend to ram. Advanced features like monitor/injection work, but when I changed the card's mac address it stopped working. I also noticed that if I then start airodump, then wireless works with new mac. ** 3 - internal mic doesn't work. tried model=acer, model=auto. Overall it seems that alsa misprograms O/B realteck codec. I talk about this later. Have same issue on my acer 5720 ** 4 - wireless led doesn't work. ath5k devs, can you fix this? ** 5 - coretemp doesn't show cpu temperature, I have seen somewhere that atom support same thermal diode as core2 and only patch to detect it is needed. Please include such path in 2.6.28 if exists. ** 6 - both card readers are missing from lspci, is this normal? ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted textures (compiz is running, and working well thought) Now I installed 2.6.28-rc3 (actually latest -git) Sadly nothing improved. I only found few regressions: * Wireless now drops connection from my WPA2 access point after few minutes. I changed temporarily its wireless protection to WEP and open mode, and both work fine, maybe this is related to periodic key exchange WPA does? * System doesn't reboot/shutdown, just hangs. Best regards, Maxim Levitsky.

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

On Sat, 2008-12-13 at 16:08 +0200, Nick Kossifidis wrote: > 2008/12/13 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > > > Hi, > > > > Indeed gpio #3 controls the led. > > I patched ath5k to use it, and led works now fine. > > > > > > Please share your patch with the rest and also post an lspci -vnn for > your card so we can handle this in a generic way This just a one liner hack diff --git a/drivers/net/wireless/ath5k/base.c b/drivers/net/wireless/ath5k/base.c index 9d2c597..a447937 100644 --- a/drivers/net/wireless/ath5k/base.c +++ b/drivers/net/wireless/ath5k/base.c @@ -2595,6 +2595,11 @@ ath5k_init_leds(struct ath5k_softc *sc) sc->led_pin = 1; sc->led_on = 1; /* active high */ } + + __set_bit(ATH_STAT_LEDSOFT, sc->status); + sc->led_pin = 3; + sc->led_on = 1; + if (!test_bit(ATH_STAT_LEDSOFT, sc->status)) goto out; lspci -vnn: 03:00.0 Ethernet controller [0200]: Atheros Communications Inc. AR242x 802.11abg Wireless PCI Express Adapter [168c:001c] (rev 01) Subsystem: Foxconn International, Inc. Device [105b:e008] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at 55200000 (64-bit, non-prefetchable) [size=64K] Capabilities: <access denied> Kernel driver in use: ath5k Kernel modules: ath5k Best regards, Maxim Levitsky

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 13 Dec '08

13 Dec '08

On Tue, 2008-11-11 at 00:12 +0200, Nick Kossifidis wrote: > 2008/11/10 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > Bob Copeland wrote: > >> On Fri, Nov 7, 2008 at 10:26 PM, Maxim Levitsky <maximlevitsky(a)gmail.com> wrote: > >>> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by > >>> default in ubuntu... > >>> wireless more or less works, but kernel log is full of backtraces. > >> > >> As you've seen, these should at least be gone now... > >> > >>> Was able to connect to my WPA2 access point. > >>> Sometimes wireless fails completely, especially after suspend to ram. > >> > >> Did you get noise calibration failures in dmesg at this point? Felix posted > >> a patch recently that handles them better. > > Yep, but they seems to be gone in latest -git too. > > Wireless work fine now, it seems like my iwl3945. > > > >> > >>> Advanced features like monitor/injection work, but when I changed the card's > >>> mac address it stopped working. > >>> I also noticed that if I then start airodump, then wireless works with new > >>> mac. > > Why it doesn't accept new mac?, can this be fixed? > > > > > >> > >>> ** 4 - wireless led doesn't work. > >>> ath5k devs, can you fix this? > >> > >> Currently all the LED code we have in there is for setting a particular gpio > >> for particular laptops (and they differ between models). We only have quirks > >> for IBM and HP, none for Acer. Also, the legacy-hal seems to have a different > >> strategy for 2425 chips. I can try to hack up some code for you to test. > >> > > > > I have seen on the web that madwifi did support the led with some gpio settings > > > > sysctl -w dev.wifi0.ledpin=3 > > sysctl -w dev.wifi0.softled=1 > > > > from https://help.ubuntu.com/community/AspireOne > > > > Best regards, > > Maxim Levitsky > > i think that some led settings are stored on EEPROM, let me check it out... Hi, Indeed gpio #3 controls the led. I patched ath5k to use it, and led works now fine. What I dislike is the way led blinks: While nether I like constant rate blinking implemented in windows driver, I don't like the fact that led blinks too fast, sometimes the blink time is so short that led doesn't show full brightness. I think there should be a minimal interval for blinking. Also I don't know what is assumed better, when led is always on, but turns off when active of vise versa? (Due to short intervals, when I set it for first option, it is almost always on, and I almost don't see the interval when it is off) Best regards, Maxim Levitsky

1 0

[alsa-devel] EMU releases 0404/0202 MAC driver stack as open source!!
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi folks. FYI - see below. Does this makes live easier to get the ALSA driver developed. THX Klaus ---Quote--- SCOTTS VALLEY, CA - December 1, 2008 - E-MU Systems today announced that for the first time it will release as open source the Macintosh driver stack for its award winning USB 2.0 audio interface products - the 0202USB, 0404USB, and TrackerPre. The Software Developer Kit will be available on the SourceForge developer's website in early December and can be found at: https://sourceforge.net/projects/zaudiodrivermac/. The software will be released under the GNU Lesser General Public License or LGPL v3, which enables developers to commercially redistribute products derived from this code base. The Software Developer Kit will include the E-MU USB 2.0 driver interface specification and necessary source code for adding new features, enhancing performance, or porting to other Operating Systems. "We are releasing this code to the public to enable programming enthusiasts the ability to experiment with and to extend the capabilities of our products, which we find being used in a variety of applications outside of the MI space proper. I'm convinced that our clever user base will come up with innovative variations not addressed by our mainstream activities," said Michael Lee, marketing manager for E-MU Systems. "I'm excited to see the results generated by the OpenSource community." ---End Quote---

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[alsa-devel] [Question] Can I open a substream in kernel space without attach to a file pointer?
by Bryan Wu 12 Dec '08

12 Dec '08

Hi Takashi, I am developing a USB gadget driver compliant to USB Audio Class Spec 2.0. So I want to open a PCM substream and do some playback of capture, then close them? I found snd_pcm_open_substream() is for opening a substream and attach it to a file. But in my application, there is no need to open a file before opening a substream. - Is there any interface for me to open a substream in kernel space without attach to a file? - How to playback and capture in kernel space, use snd_pcm_lib_write and snd_pcm_lib_read? - How to get the snd_pcm_hardware struct from low level driver, because I have to get the hardware configuration of the snd pcm device? And I am reading the code of OSS emulator in ALSA. It provides some info about the kernel space sound card programming. Thanks a lot -Bryan

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[alsa-devel] Macbook Air 2.1 - No sound on main speakers
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi there. Fowarded from my "Ubuntu manual upgrade thread": Another problem with Realtek. No sound on mainspeakers on MACBOOK Air 2.1. A known issue?? See also the different aplay and codec IDs! Klaus Quote: ----------------------- No sound on main internal speaker. Headphones do work fine. Macbook Air 2,1 Ubuntu Intrepid server Kernel 2.6.27-9 Alsa Driver version 1.0.18a GIT snapshot from 12/4/08 Sound card nVidia MCP79 (device id 10de:0ac0) aplay -l: ALC885 /proc/asound/card0/codec#0 CODEC ALC889a lspci -nv 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb79 ------------------------

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[alsa-devel] HP Pavillion TX2000z - Headphonejack not working
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi folks. Another problem now with HP and realtek AC861VD. The headphone-jack is not working. Known issue? THX \Klaus QUOTE: -------------------------- drivers 1.0.18a HP Pavillion tx2000z, Realtek AC861VD, Alsa version 1.0.18a. Running Ubuntu Studio Intrepid 8.10 Kernel 2.6.27-9-generic options snd-hda-intel model=hp kqian@kkqian:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC861VD Digital [ALC861VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 kqian@kkqian:~$ cat /proc/asound/card0/codec#* | grep Codec Codec: Realtek ALC861-VD Codec: Motorola Si3054 Oh, I should also add, headphones do not show up in volume control. I have checked the switches, as well as the main interface. The only things that are there are: Master, PCM, Front, Front Mic, Front Mic Boost, Microphone, Mic Boost, Speaker, Capture, IEC958, IEC958 Default PCM, Caller ID, Off-hook, Input Source, All of them are checked, and none of them are muted.

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[alsa-devel] More notes on HD-audio driver
by Takashi Iwai 11 Dec '08

11 Dec '08

Hi, per popular demand, I wrote a brief document more about HD-audio driver. It's now found in sound git tree as Documentation/sound/alsa/HD-Audio.txt. The git tree is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git You can get the text from web: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=blob_plain;… A PDF file is found at: http://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf Comments, corrections and additions are appreciated. thanks, Takashi

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[alsa-devel] [PATCH 0/2] ASoC updates
by Mark Brown 11 Dec '08

11 Dec '08

The following changes since commit 6de45d5d776d2a7e7a9adc8ea49d37fe1bd45fb2: Takashi Iwai (1): ALSA: ASoC - Fix DAI registration in s3c2443-ac97.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (1): ASoC: Fix variable name for Blackfin I2S DAI Stanley.Miao (1): ASoC: TWL4030: hands-free start-up sequence. sound/soc/blackfin/bf5xx-i2s.c | 4 ++-- sound/soc/codecs/twl4030.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 6 ++++++ 3 files changed, 38 insertions(+), 6 deletions(-)

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[alsa-devel] [PATCH] TWL4030: Add event callback to enable hands-free.
by Stanley.Miao 11 Dec '08

11 Dec '08

A special sequence is required to enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao(a)windriver.com> --- sound/soc/codecs/twl4030.c | 33 +++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 6 ++++++ 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 71ab534..86d6553 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -325,6 +325,29 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, return ret; } +static int handsfree_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; + unsigned char hs_ctl; + + + hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + + if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + hs_ctl |= TWL4030_HF_CTL_RAMP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_LOOP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_HB_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + } else { + twl4030_write(w->codec, e->reg, 0x0); + } + + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -807,10 +830,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitr_control), /* HandsfreeL/R */ - SND_SOC_DAPM_MUX("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, - &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_MUX("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, - &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, + &twl4030_dapm_handsfreel_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, + &twl4030_dapm_handsfreer_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index a2065d4..54615c7 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -191,6 +191,12 @@ #define TWL4030_RAMP_DELAY_2581MS 0x1C #define TWL4030_RAMP_EN 0x02 +/* HFL_CTL (0x29, 0x2A) Fields */ +#define TWL4030_HF_CTL_HB_EN 0x04 +#define TWL4030_HF_CTL_LOOP_EN 0x08 +#define TWL4030_HF_CTL_RAMP_EN 0x10 +#define TWL4030_HF_CTL_REF_EN 0x20 + /* APLL_CTL (0x3A) Fields */ #define TWL4030_APLL_EN 0x10 -- 1.5.6.3

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[alsa-devel] ASoC: asound.conf
by naveen krishna ch 11 Dec '08

11 Dec '08

Hi all, I have a custom built ASoC CODEC driver for twl4030. I am integrating my kernel (ASoC driver) with the application. The application expects a /etc/asound.conf file is there any utility to build or restore the asound.conf for my ASoC driver. I have checked the "alsactl" which can give me the state/controls information. Do i need to make a asound.conf file.If so, can any one help my with some information (a sample asound.conf) with details of the driver in it. Sorry, if this the wrong place to raise this issue. -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] ASoC: asound.conf
by naveen krishna ch 11 Dec '08

11 Dec '08

Hi all, I have a custom built ASoC CODEC driver for twl4030. I am integrating my kernel (ASoC driver) with the application. The application expects a /etc/asound.conf file is there any utility to build or restore the asound.conf for my ASoC driver. I have checked the "alsactl" which can give me the state/controls information. Do i need to make a asound.conf file.If so, can any one help my with some information (a sample asound.conf) with details of the driver in it. Sorry, if this the wrong place to raise this issue. -- Thanks, (: Naveen Krishna Ch :)

1 0

[alsa-devel] 2.6.26.[6|7]-rt11, alsa rawmidi, seq hang
by Fernando Lopez-Lezcano 11 Dec '08

11 Dec '08

(starting a new thread with the hope of being more precise after a lot of testing and rebooting) I'm seeing a realtime patch related hard hang in the kernel alsa subsystem (MIDI input/output). In a nutshell: - alsa rawmidi works (ie: "rawmidi -v -i hw:0" outputs a stream of messages when pointed to a midi capable card that has an external keyboard connected). - the alsa sequencer interface works (ie: aplaymidi connected to aseqdump transfers data just fine). - BOTH combined do NOT work (ie: use aconnect to connect the port that corresponds to the external midi interface to aseqdump: aseqdump hangs forever after transferring the first message and the only way out is a reboot). Just tested this on 2.6.26.7 + rt11, Gina3G soundcard with an external MIDI keyboard sending messages. This is not specific to the Gina card, I have tested also USB midi interfaces and an RME hdsp card with a Multiface interface. 2.6.24.7 + rt21 works fine on the same hardware combinations. The standard Fedora 9 2.6.26.x kernel works fine as well. I'm attaching the kernel configuration for the kernel I'm currently testing plus a dmesg dump of the whole boot process (rt kernel hackers can find there some additional sleeping in wrong context warning messages there), including the output of a "echo t >/proc/sysrq-trigger" that should show where aseqdump currently hangs (or so I think). Hopefully this will narrow the area affected enough so that a solution can be found or more debugging info can be requested. Currently 2.6.26.x-rtx is unusable for musical purposes (at least when you have to interact with external midi controllers). Thanks for any help... -- Fernando

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Re: [alsa-devel] More notes on HD-audio driver
by kls.schlz＠googlemail.com 11 Dec '08

11 Dec '08

If you prepare a Wiki for the HD-Audio, don't forget to merge somehow ALSA-Configuration.txt. IMO this goes together. On Dec 11, 2008 7:24am, Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 10 Dec 2008 14:45:52 -0700, > > stan wrote: > > > > > > Takashi Iwai wrote: > > > > Hi, > > > > > > > > per popular demand, I wrote a brief document more about HD-audio driver. > > > > It's now found in sound git tree as Documentation/sound/alsa/HD-Audio.txt. > > > > The git tree is: > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > > > > > You can get the text from web: > > > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=blob_plain;… > > > > > > > > A PDF file is found at: > > > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf > > > > > > Thank you for this. I found it informative and helpful. > > > > > > Maybe it could be placed on the wiki or sent to the > > > alsa-user list once a month as "how to troubleshoot hd audio"? > > > > Putting Wiki is a good idea. I'm not sure whether it'd be better to > > put the text into Wiki or to put a link to the above URL, though... > > > > > > Takashi > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

2 1

[alsa-devel] Idea about the laptop speakers
by Alexander E. Patrakov 11 Dec '08

11 Dec '08

Hello developers, some time ago, people complained that the sound came out from both the laptop speakers and headphones. While in many cases this is not desirable, I think this effect (when configurable via opening a different PCM) can be useful in the following situation. Suppose that you have a laptop as your main working machine. The headset is always connected, but you are not always near the laptop (e.g., think better if you walk in the room). Then, you configure (by telling it to open different devices) your SIP softphone to play voice through headphones only, but play ringtones through both headphones and speakers. Thus, you can walk in the room without being afraid to miss a call. Is the above proposal implementable? -- Alexander E. Patrakov

1 0

[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit f73f2a6a23e34de9cca9672f727694e5af00e6c7: Takashi Iwai (1): ALSA: ASoC - Fix symbol conflicts in omac-mcbsp.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (5): ASoC: Work around warnings from some build environments ASoC: Add codec registration API ASoC: Convert WM8900 to do more work at I2C probe time ASoC: Convert WM8903 driver to register at I2C probe time ASoC: Stop WM8903 SYSCLK when suspending Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs include/sound/soc.h | 5 + sound/soc/codecs/twl4030.c | 27 +++--- sound/soc/codecs/wm8900.c | 159 +++++++++++++++-------------- sound/soc/codecs/wm8900.h | 7 -- sound/soc/codecs/wm8903.c | 241 +++++++++++++++++++------------------------- sound/soc/codecs/wm8903.h | 5 - sound/soc/soc-core.c | 43 ++++++++ 7 files changed, 247 insertions(+), 240 deletions(-)

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[alsa-devel] [PATCH] ALSA: Add support for mechanical jack insertion
by Mark Brown 10 Dec '08

10 Dec '08

Some systems support both mechanical and electrical jack detection, allowing them to report that a jack is physically present but does not have any functioning connections. Add a new jack type for these, allowing user space to report faulty connections. Thanks to Guillem Jover for the suggestion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/input.h | 1 + include/sound/jack.h | 1 + sound/core/jack.c | 6 ++++++ 3 files changed, 8 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index 1a958cd..9a6355f 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -660,6 +660,7 @@ struct input_absinfo { #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ +#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 7cb25f4..2e0315c 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -36,6 +36,7 @@ enum snd_jack_types { SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..7ca905c 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -108,6 +108,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); + if (type & SND_JACK_MECHANICAL) + input_set_capability(jack->input_dev, EV_SW, + SW_JACK_PHYSICAL_INSERT); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); + if (jack->type & SND_JACK_MECHANICAL) + input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, + status & SND_JACK_MECHANICAL); input_sync(jack->input_dev); } -- 1.5.6.5

2 1

[alsa-devel] Improvements on aloop driver
by Takashi Iwai 10 Dec '08

10 Dec '08

Hi, as some people have asked about aloop bugs and developments, I started working on it a bit. The fact is that aloop is buggy per design in two points: - Sharing the same buffer for both playback and capture doesn't work. The playback app may change the "available" buffer area at any time. It can update the last played period very quickly. OTOH, the capture needs to read the last period. So, it obviously conflicts. - We needs to synchronize both playback and capture streams. In the current implementation, both streams are updated individually. For example, the capture can start in the middle of the playback period. This results in the partial available samples. So, I wrote the code and got almost another shape of the driver. The result is found in my alsa-driver-build git tree git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git and the snapshot tarball ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Let me know whether this works for you. Once after confirmed to work well enough, we should push this to the upstream -- or abandon if the driver is conceptually wrong... thanks, Takashi

1 0

[alsa-devel] [PATCH 0/3] ASoC: TWL4030: Fixes and small cleanup
by Peter Ujfalusi 10 Dec '08

10 Dec '08

Hello, The CarkitL/R DAPM output somehow got lost, the first patch adds it. Really small cleanup for line wrap in the mux switch texts. DAC names changed in order to avoid confusion with the mux switch names. --- Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs sound/soc/codecs/twl4030.c | 27 +++++++++++++-------------- 1 files changed, 13 insertions(+), 14 deletions(-)

2 4

[alsa-devel] Ask a question about WM8976
by peter meng 10 Dec '08

10 Dec '08

Hi, I'm a newbie for alsa and using wm8976 first time . I found the driver of wm8976 comments the contents of wm8976_dapm_widgets . 1. Is this mean all widget in wm8976 does not have power management function ? 2. If wm8976 does not add widget then the process of 'set up audio path map ' will fails , is this correct ? ------------------------------------------------------------------- // TODO Widgets static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { #if 0 //SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), //SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, &wm8976_speaker_mixer_controls[0], ARRAY_SIZE(wm8976_speaker_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, &wm8976_mono_mixer_controls[0], ARRAY_SIZE(wm8976_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, &wm8976_aux_boost_controls, 1), SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, &wm8976_mic_boost_controls, 1), SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, &wm8976_capture_boost_controls), SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), #endif }; static int wm8976_add_widgets(struct snd_soc_codec *codec) { int i; for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) { snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]); } /* set up audio path map */ for(i = 0; audio_map[i][0] != NULL; i++) { snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); } snd_soc_dapm_new_widgets(codec); return 0; } ----------------------------------------------- Thanks advanced . Best Regards Peter Meng

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Re: [alsa-devel] Warnings in wm8903.c (Re: Merged upstream to topic/asoc branch)
by Mark Brown 10 Dec '08

10 Dec '08

On Wed, Dec 10, 2008 at 08:26:56AM +0100, Takashi Iwai wrote: > Hmm, I got the warnings again on a x86-64 with gcc-4.3.2 randconfig > test build again. > The config is attached below. OK, I've managed to reproduce this with that .config. I'll push a patch later today which adds return statements to silence the warning, meaning that any genuine issues won't be suppressed.

1 0

[alsa-devel] [PATCH 00/18] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit 28a1d869560a49d960ba2a3b450ec965712e5560: Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (9): ASoC: Add card registration API ASoC: Add DAI registration API ASoC: Register platform DAIs ASoC: Add platform registration API ASoC: Register platform drivers ASoC: Register non-AC97 codec DAIs ASoC: Initial framework for dynamic card instantiation ASoC: Wait for non-AC97 codec DAIs before instantiating ASoC: Convert WM8900 to allow registration by machine code Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down include/sound/soc-dai.h | 8 + include/sound/soc.h | 10 ++ sound/soc/atmel/atmel-pcm.c | 12 ++ sound/soc/atmel/atmel_ssc_dai.c | 11 ++ sound/soc/au1x/dbdma2.c | 3 +- sound/soc/au1x/psc-ac97.c | 3 +- sound/soc/au1x/psc-i2s.c | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-ac97.c | 12 ++ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-i2s.c | 12 ++ sound/soc/codecs/ad73311.c | 12 ++ sound/soc/codecs/ak4535.c | 12 ++ sound/soc/codecs/cs4270.c | 12 ++ sound/soc/codecs/pcm3008.c | 12 ++ sound/soc/codecs/ssm2602.c | 12 ++ sound/soc/codecs/tlv320aic23.c | 12 ++ sound/soc/codecs/tlv320aic26.c | 15 ++- sound/soc/codecs/tlv320aic3x.c | 12 ++ sound/soc/codecs/twl4030.c | 277 +++++++++++++++++++++++++++++++-- sound/soc/codecs/uda134x.c | 12 ++ sound/soc/codecs/uda1380.c | 12 ++ sound/soc/codecs/wm8510.c | 12 ++ sound/soc/codecs/wm8580.c | 12 ++ sound/soc/codecs/wm8728.c | 12 ++ sound/soc/codecs/wm8731.c | 12 ++ sound/soc/codecs/wm8750.c | 12 ++ sound/soc/codecs/wm8753.c | 12 ++ sound/soc/codecs/wm8900.c | 105 ++++--------- sound/soc/codecs/wm8903.c | 12 ++ sound/soc/codecs/wm8971.c | 12 ++ sound/soc/codecs/wm8990.c | 12 ++ sound/soc/davinci/davinci-i2s.c | 12 ++ sound/soc/davinci/davinci-pcm.c | 12 ++ sound/soc/fsl/fsl_dma.c | 12 ++ sound/soc/fsl/fsl_ssi.c | 10 ++ sound/soc/fsl/mpc5200_psc_i2s.c | 4 + sound/soc/omap/omap-mcbsp.c | 13 ++ sound/soc/omap/omap-pcm.c | 12 ++ sound/soc/pxa/pxa-ssp.c | 12 ++ sound/soc/pxa/pxa2xx-ac97.c | 12 ++ sound/soc/pxa/pxa2xx-i2s.c | 13 ++- sound/soc/pxa/pxa2xx-pcm.c | 12 ++ sound/soc/s3c24xx/s3c2412-i2s.c | 13 ++ sound/soc/s3c24xx/s3c2443-ac97.c | 13 ++ sound/soc/s3c24xx/s3c24xx-i2s.c | 12 ++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 ++ sound/soc/sh/dma-sh7760.c | 12 ++ sound/soc/sh/hac.c | 12 ++ sound/soc/sh/ssi.c | 12 ++ sound/soc/soc-core.c | 291 +++++++++++++++++++++++++++++++++-- 51 files changed, 1102 insertions(+), 110 deletions(-)

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[alsa-devel] No sound from laptop speakers (Resolved)
by Manilal K M 10 Dec '08

10 Dec '08

Hello everybody, Recently I have installed Fedora 10 on my Compaq CQ45 laptop. But unfortunately there was no sound from the laptop speakers even though the headphones worked perfectly. I did some google search but it didn't helped. Later, on a discussion at #alsa(freenode.net), one of the IRC member(nickname: copper) gave some clue and finally found that I have to upgrade the alsa driver and pass the model id(dell-m4-1) as a parameter in modprobe.conf. Here are the results generated by alsa-info.sh before and after upgrading to the latest alsa(1.0.18a). http://www.alsa-project.org/db/?f=9a29eac2731b3de1c17183d6c43aac3fa11942c9 http://www.alsa-project.org/db/?f=9a29eac2731b3de1c17183d6c43aac3fa11942c9 The option I have passed to /etc/modprobe.conf/alsa-base is : options snd-hda-intel model=dell-m4-1 After reboot, I was able to hear sound from laptop speakers. -- Manilal K M : മണിലാല്‍ കെ എം. http://libregeek.blogspot.com

2 2

[alsa-devel] [PATCH 2/2] ASoC: TWL4030: Small cleanup
by Peter Ujfalusi 10 Dec '08

10 Dec '08

The mux switch related texts fits to on line, no need to wrap them. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 9 +++------ 1 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df9f2ce..e3a7115 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,8 +192,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* Earpiece */ static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR1"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR1"}; static const struct soc_enum twl4030_earpiece_enum = SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, @@ -205,8 +204,7 @@ SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); /* PreDrive Left */ static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR2"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR2"}; static const struct soc_enum twl4030_predrivel_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, @@ -218,8 +216,7 @@ SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); /* PreDrive Right */ static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "Invalid", - "DACL2"}; + {"Off", "DACR1", "DACR2", "Invalid", "DACL2"}; static const struct soc_enum twl4030_predriver_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, -- 1.6.0.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: TWL4030: Add missing Carkit output
by Peter Ujfalusi 10 Dec '08

10 Dec '08

SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 373daa4..df9f2ce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -761,6 +761,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PREDRIVER"), SND_SOC_DAPM_OUTPUT("HSOL"), SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("CARKITL"), + SND_SOC_DAPM_OUTPUT("CARKITR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), -- 1.6.0.4

1 0

[alsa-devel] How to write a PXA CPU
by Juan Pablo Gómez 09 Dec '08

09 Dec '08

Hi, I'm looking for documentation that explains me how to write a PXA document. It's for a NIOS 2 processor in an altera De2 board. -- Regards, Juan Pablo Muchas gracias. Atentamente, Juan Pablo Gómez

2 3

[alsa-devel] [PATCH 0/9] ASoC: TWL4030: DAPM implementation for outputs (playback) - try3
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, The whole series again, altrough 1 and 2 has been accepted already. This series of patches (try3) adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. Remove the Headset output gain modifications from the twl4030_power_up and twl4030_power_down function, since they are not needed. Changes since try1 (comments addressed): - Removed the " Switch" from the option texts and also from the intercon. --- Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down sound/soc/codecs/twl4030.c | 263 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 248 insertions(+), 15 deletions(-)

2 10

[alsa-devel] [PATCH 0/9] ASoC: TWL4030: DAPM implementation for outputs (playback) - try2
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, This series of patches (try2) adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. Remove the Headset output gain modifications from the twl4030_power_up and twl4030_power_down function, since they are not needed. Changes since try1 (comments addressed): - erapiece typo - too fast/slow finger combination ;) - Removed the " Switch" from the option texts - rename the confusing _controls to _control - Analog PGA control reworked: - SND_SOC_DAPM_PGA for power switch - Normal control stereo (user control) for DA enable/disable (mute/unmute): - "DAC1 Analog Playback Volume" + "DAC1 Analog Playback Switch" - "DAC2 Analog Playback Volume" + "DAC2 Analog Playback Switch" --- Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down sound/soc/codecs/twl4030.c | 263 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 248 insertions(+), 15 deletions(-)

2 13

[alsa-devel] [PATCH 0/8] ASoC: TWL4030: DAPM implementation for outputs (playback)
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, This series of patches adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. --- Peter Ujfalusi (8): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs sound/soc/codecs/twl4030.c | 248 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 243 insertions(+), 5 deletions(-)

2 20

[alsa-devel] ASoC: hook for codec control updates and clock controls
by Daniel Mack 08 Dec '08

08 Dec '08

Hi, Codecs like the tlv320aic3x family have soft volume controls to smoothly drive up and down mixer elements in order to prevent clicks and pops. In fact, most of the controls on this chip are implemented as such. For the mixers to work, however, the I2S input clock is needed which is currently only true when the DAC/ACD elements are active but not when the chip is just operating as an analog mixer. To allow this functionality, the I2S clock must be present for a short period after each volume control update to the codec and disabled again after some milliseconds. It should, however, not be enabled if it's already running and of course, it must not be disabled if the ADC/DAC is still running. Adding a hook to struct snd_soc_codec which is called from snd_soc_update_bits() isn't a big deal, but the problem is the abstraction layers in this case - for good reasons, the cpu_dai is not accessible from the codec dai. Or did I miss a link? Is there a sane way to address this problem? Thanks and best regards, Daniel

3 9

[alsa-devel] [patch] alsa-utils fix for intl support
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Feel free to comment on the included patch. I had to add it because aseqnet would not link without -lintl. -- Best regards, Hans-Christian Egtvedt

2 5

[alsa-devel] [patch] alsa-utils check if __USE_BSD is defined before compiling "BSD functions"
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Another bug/issue I tripped over when compiling alsa-utils in an environment using uClibc to supply the C library functions. Here I have enabled some old BSD style functions. The attached patch will honor them if they are enabled. Without this patch I get a redefined error during compile. -- Best regards, Hans-Christian Egtvedt

2 3

[alsa-devel] [PATCH 0/7] ASoC updates
by Mark Brown 08 Dec '08

08 Dec '08

The following changes since commit 682d5874f3d654b5d13d9b8dd56b9e05cfadd01b: Takashi Iwai (1): ALSA: ASoC: Fix old-style trigger callback in s3c2443-ac97.c are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Rename snd_soc_register_card() to snd_soc_init_card() ASoC: Annotate core removal function ASoC: Remove in-code changelog from AD73311 driver ASoC: Don't free static data in WM9713 Peter Ujfalusi (3): ASoC: TWL4030: Change the capture volume control to TLV ASoC: TWL4030: Change the common playback volume controls ASoC: TWL4030: Add volume controls for outputs include/sound/soc.h | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 5 +-- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/twl4030.c | 74 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 3 +- sound/soc/soc-core.c | 8 ++-- 27 files changed, 91 insertions(+), 45 deletions(-)

4 12

[alsa-devel] HDA: CLFE DAC on ALC889 (Acer Aspire 8930G)
by Hector Martin 07 Dec '08

07 Dec '08

I've been trying to get ALSA working with this laptop. So far the attached patch adds rear support to the existing 4930g model, and tries to add CLFE support. However, I haven't been able to get CLFE to work. I don't seem to get any audio through DACs/mixers other than the first two (front and rear). I can assign the CLFE output (pin 0x16) to the front or rear DAC and hear the front audio through the CLFE speakers. I can also force the front DAC (0x02) or rear DAC (0x03) to use the CLFE stream and hear it through the front or rear speakers, so the audio is getting to the codec. But either DACs 0x04 and 0x05 aren't working, or their associated mixers (0x0e and 0x0f) aren't passing through the audio, or the OUT pins don't like selections above 1. I've checked that they're unmuted and that the volumes are turned up on both. I've also tried the 0x25->0x26 DAC/mixer auxiliary pathway, which doesn't work either. Here's what the relevant NIDs correspond to: 0x02 - (normally) Front DAC 0x03 - (normally) Rear DAC 0x04 - (normally) CLFE DAC 0x05 - (presumably) side DAC (no side speakers) 0x0c - (normally) Front mixer 0x0d - (normally) Rear mixer 0x0e - (normally) CLFE mixer 0x0f - (presumably) side mixer 0x14 - Front speakers pin 0x15 - Headphones pin (also EAPD for speaker amp) 0x16 - CLFE speakers pin 0x1b - Rear speakers pin Here's what the codec graph looks like: http://marcansoft.com/transf/codec.png Any ideas? I'm pretty stumped here. Are there any other magic bits to enable the remaining DACs or mixers? -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

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Re: [alsa-devel] [Swfdec] Can't use Mozilla plugin or play video with 0.8.2 / 0.9.2
by Benjamin Otte 07 Dec '08

07 Dec '08

I think I know what this is. But I spent the best part of my breakfast wondering about it. I think using Pulse Audio might work around this problem, but I think it's a driver, configuration issue on the ALSA side or a misunderstanding of the API on my part, which is why I'm sending this to the ALSA list. So here's the problem: Swfdec uses ALSA in non-blocking mode and polls its file descriptor(s). Whenever the audio queue becomes empty enough that it's useful to render more audio, the file descriptor wakes up from the poll. This setting is apparently configurable using various hw params, however, Swfdec does not configure it and has been relying on the fact that the sound device picks proper defaults, thinking that a sound card with a small buffer will likely want to wake up more often than something like Pulse Audio's new design, which keeps a 2+ second buffer for the audio. It seems though that your sound card wakes up roughly whenever a new sample can be rendered, which is 44100 times per second (or more, as from your profile it looks like ALSA resamples the audio). This causes the whole rendering code inside Swfdec to run for every sample, which is so much overhead that it kills all your performance. My question to you ALSA people now is: Who's responsible for ensuring that the number of wakeups is sane? Should I do some configuring of the audio device and if so, what kind of configuration, or should ALSA be able to pick reasonable defaults for me? Cheers, Benjamin On Sun, Dec 7, 2008 at 2:46 AM, Jesse Zbikowski <embeddedlinuxguy(a)gmail.com> wrote: > Hi Riccardo, > > On Fri, Dec 5, 2008 at 2:32 PM, Riccardo Magliocchetti > <riccardo.magliocchetti(a)gmail.com> wrote: >> Give it a run with oprofile, this is my opreport > > Thanks for this data. It looks like I may have an audio bottleneck. > opreport -t3 shows > > 31.5811 no-vmlinux > 14.4701 libswfdec-0.9.so.3.0.0 > 8.3210 libasound_module_rate_speexrate.so > 6.0088 libc-2.8.90.so > 4.7060 libpixman-1.so.0.12.0 > 3.6615 python2.5 > 3.2831 libcairo.so.2.10800.0 > > If I just look at the libraries I see the top libswfdec routines are > all audio related: > > 14.1497 libswfdec-0.9.so.3.0.0 swfdec_sound_buffer_get_n_samples > 2.1666 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_channels > 1.9369 libswfdec-0.9.so.3.0.0 swfdec_audio_format_new > 1.7904 libswfdec-0.9.so.3.0.0 swfdec_audio_flv_render > 1.3205 libswfdec-0.9.so.3.0.0 __i686.get_pc_thunk.bx > 1.2073 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_granularity > > I could try using the PulseAudio backend instead of ALSA. > >> You can try to improve your performance by using latest stable version of >> cairo and pixman > > OK, I am currently using the latest Ubuntu versions which are a bit > behind latest stable. I will see if this makes a difference. > > Package Ibex Latest > xorg-xserver 1.5.2 1.5.3 > cairo 1.8.0 1.8.4 > pixman 0.12 0.13.2 > >> play a bit with your X video driver config ie. exa vs xaa. > > I did not see libexa.so in my oprofile output although it is > installed, so I will make sure this is enabled. > > Thanks again, > Jesse Z. > _______________________________________________ > Swfdec mailing list > Swfdec(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/swfdec >

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Re: [alsa-devel] Status of the aloop loopback driver
by Takashi Iwai 05 Dec '08

05 Dec '08

At Fri, 5 Dec 2008 17:11:07 +0000, Andy Whitcroft wrote: > > I am interesting in finding out the status of, and plans for the aloop > loopback driver. I note that this has been about for a while and has > not been pushed up to the mainline kernel. Are you intending on > pushing that up or is it planning on being in the alsa-drivers tarball > for the forseeable future? It's still buggy, unfortunately, due to the current design. The timing of capture and playback isn't synchronized, so you'll get often noisy outputs. I've worked on this driver yesterday, just did some clean up works. Hopefully I'll be able to fix the issue sometime later... thanks, Takashi

1 0

[alsa-devel] [PATCH] tlv320aic3x: control additions and cleanups
by Daniel Mack 05 Dec '08

05 Dec '08

Some minor updates for sound/soc/codecs/tlv320aic3x.c: - removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - split "Line Playback Switch" into "LineL Playback Switch" and "LineR Playback Switch" - split "Line PGA Bypass Playback Volume" into "LineL Left PGA Bypass Playback Volume" and "LineR Right PGA Bypass Playback Volume" - split "Line Line2 Bypass Playback Volume" into "LineL Line2 Bypass Playback Volume" and "LineR Line2 Bypass Playback Volume" - Added "HP Right PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Uli Luckas <u.luckas(a)road.de> --- sound/soc/codecs/tlv320aic3x.c | 24 ++++++++++++++++-------- 1 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 537bbbf..7cdaea3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -253,16 +253,22 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, - 0x01, 0), - SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - PGAR_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, + 0, 0x7f, 1), + SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -270,8 +276,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, + PGAR_2_HPROUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, @@ -281,7 +289,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

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[alsa-devel] [PATCH] tlv320aic3x.c: controller naming fixes
by Daniel Mack 05 Dec '08

05 Dec '08

- removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - renamed "HPL PGA Bypass Playback Volume" to "HPL Left PGA Bypass Playback Volume" - renamed "HPR PGA Bypass Playback Volume" to "HPR Left PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- This is the second part of the patch I sent yesterday, plus the naming fixes recently discussed on the list. sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a12dc3..081f7c4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -268,7 +268,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -276,20 +276,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + SOC_SINGLE("HPL Left PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + SOC_SINGLE("HPR Left PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

1 0

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 05 Dec '08

05 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 321 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..bd91594 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,311 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(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 + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + } else { + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + mdelay(1); + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + } + + return 0; +} + +/* + * Audio paths on Pandora board: + * + * |O| ---> PCM DAC +-> AMP -> Headphone Jack + * |M| A +--------> Line Out + * |A| <~~clk~~+ + * |P| <--- TWL4030 <--------- Line In and MICs + */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, omap3pandora_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Amplifier", NULL, "PCM DAC"}, + {"Line Out", NULL, "PCM DAC"}, + {"Headphone Jack", NULL, "Headphone Amplifier"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .platform = &omap_soc_platform, + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + pr_err(PREFIX "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 3

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 04 Dec '08

04 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- I'm not sure if splitting DAPM and widgets was good idea, I can adjust if needed. The map is also not quite right, it says headphone jack is connected to TWL4030 codec OUTL and OUTR, but that is not really the case. It is connected to external DAC, and DAC is connected to OMAP. However DAC uses clock from TWL4030, so TWL needs to be active. I've basically done that connection to get event callback, where I can switch DAC and headphone amp on/off. sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 293 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 303 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..4f92877 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,293 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(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 + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + + return 0; +} + +/* Machine DAPM widgets */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", omap3pandora_hp_event), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Jack", NULL, "OUTL"}, + {"Headphone Jack", NULL, "OUTR"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + printk(KERN_ERR "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 04 Dec '08

04 Dec '08

The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2: Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Push debugfs files out of the snd_soc_device structure ASoC: Remove device from platform suspend and resume operations ASoC: Remove platform device from DAI suspend and resume operations ASoC: Remove obsolete declaration of struct snd_soc_clock_info include/sound/soc-dai.h | 6 +-- include/sound/soc.h | 15 +++---- sound/soc/atmel/atmel-pcm.c | 6 +-- sound/soc/atmel/atmel_ssc_dai.c | 6 +-- sound/soc/au1x/psc-ac97.c | 6 +-- sound/soc/au1x/psc-i2s.c | 6 +-- sound/soc/blackfin/bf5xx-ac97.c | 6 +-- sound/soc/blackfin/bf5xx-i2s.c | 6 +-- sound/soc/pxa/pxa-ssp.c | 6 +-- sound/soc/pxa/pxa2xx-ac97.c | 6 +-- sound/soc/pxa/pxa2xx-i2s.c | 6 +-- sound/soc/s3c24xx/s3c2412-i2s.c | 16 +++---- sound/soc/s3c24xx/s3c24xx-i2s.c | 6 +-- sound/soc/soc-core.c | 85 +++++++++++++++++++++----------------- 14 files changed, 83 insertions(+), 99 deletions(-)

2 5

[alsa-devel] question for the all-knowing - firewire audio hardware
by S. Aguinaga 04 Dec '08

04 Dec '08

Hello! I have a couple of machines connected to FireWire Motu devices (running OS X). We want to develop some new apps on linux and the only info I've found is: http://www.olafchrist.de/ieee1394/index.php, but its rather old and single device specific. Are there any other solutions available? Or are there other devices (firewire) that are supported by the ALSA project? Any info on this will be great!! // Sal

2 3

Re: [alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Resubmiited with full name. Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by Takashi Iwai 04 Dec '08

04 Dec '08

At Wed, 3 Dec 2008 21:11:22 +0000, Scott Waye wrote: > > > Signed-off-by: Scott Waye <scott(a)waye.co.uk> Applied now. Thanks. Takashi > > diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c > index 1360d54..b78c8f2 100644 > --- a/pci/hda/patch_nvhdmi.c > +++ b/pci/hda/patch_nvhdmi.c > @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) > struct hda_codec_preset snd_hda_preset_nvhdmi[] = { > { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, > { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, > + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, > {} /* terminator */ > }; > -- > 1.5.6 >

1 0

[alsa-devel] [PATCH] ALSA: Reduce stall detection timeout in riptide.c
by nokos＠gmx.net 03 Dec '08

03 Dec '08

Reduce the command timeout to 0.5sec. Should be enough to allow a working command interface but removes a RCU stall and slow resume on some revisions where the AC97 revision detection stalls in resume. Signed-off-by: Peter Gruber <nokos(a)gmx.net> --- diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e9f0706..d0ccfc6 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -172,7 +172,7 @@ #define MAX_WRITE_RETRY 10 /* cmd interface limits */ #define MAX_ERROR_COUNT 10 -#define CMDIF_TIMEOUT 500000 +#define CMDIF_TIMEOUT 50000 #define RESET_TRIES 5 #define READ_PORT_ULONG(p) inl((unsigned long)&(p))

2 1

[alsa-devel] [PATCH 0/5] ASoC updates
by Mark Brown 03 Dec '08

03 Dec '08

The following changes since commit 0ecfe7987855d21c2a89ffe003ddf0ee11b42d47: Mark Brown (1): ASoC: Don't free static data in WM9713 are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support Grazvydas Ignotas (1): ASoC: TWL4030: Add input selection and gain controls Mark Brown (3): ASoC: Push workqueue data into snd_soc_card ASoC: Push platform registration down into the card ASoC: Fix WM8903 right mixer bypass path include/sound/soc.h | 9 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 31 ++++++- sound/soc/codecs/tlv320aic3x.h | 46 +++++++++- sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 +++ sound/soc/codecs/wm8903.c | 8 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/soc-of-simple.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-core.c | 77 +++++++++------ 36 files changed, 348 insertions(+), 74 deletions(-)

3 8

Re: [alsa-devel] Gateway MX3042 Laptop. No sound
by Takashi Iwai 03 Dec '08

03 Dec '08

At Wed, 3 Dec 2008 06:39:24 -0600, Michael Satterwhite wrote: > > On Wednesday 03 December 2008 01:02:07 am you wrote: > > At Wed, 3 Dec 2008 00:12:54 +0100, > > > > Nigel Henry wrote: > > > I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, > > > which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but > > > no sounds. > > > > > > The output from running alsa-info.sh is below. > > > > Try my sound.git tree and sound-unstable.git tree. > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git > > > > The snapshot tarballs are found in > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > > I'll be happy to (and *DO* thank you for your help!). I'm not sure what this > means, however. Are these drivers, a kernel ... how do I try them? It's a linux kernel git tree. Ask other Ubuntu guys how to build your kernel. Or, use the alsa-driver snapshot tarball in the ftp directory above. You can install it just like the normal alsa-driver release tarball. Takashi

1 0

[alsa-devel] [PATCH] tlv320aic3x: headset/button press support
by Daniel Mack 03 Dec '08

03 Dec '08

Hi, This patch adds a function to setup headset and button press detection modes. Also, aic3x_headset_detected() returns the value from the current state's register rather than the interrupt cause flag so it works without an interrupt as well. Best regards, Daniel

3 5

[alsa-devel] [PATCH] ASoC: TWL4030: Add input selection and gain controls
by Grazvydas Ignotas 03 Dec '08

03 Dec '08

The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- Compared to my previous patch, I've changed "Microphone Boost" name to "Input Boost Volume", as it affects line inputs too. I thought that hiding this control for line-in was not a good idea, as it might still be useful for weak sources. "Capture" was also changed to "Input", as inputs can be also routed to outputs, not only recorded. sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 ++++ 2 files changed, 190 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120..3c9fdf2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,9 +46,9 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ - 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICR (0x6) */ + 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ 0x0c, /* REG_ATXL1PGA (0xA) */ @@ -347,6 +347,162 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + result++; + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias, avadc_ctl; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 2: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 3: + anamicl |= TWL4030_AUXL_EN; + break; + case 4: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + break; + } + + /* If some input is selected, enable amp and ADC */ + if (value != 0) { + anamicl |= TWL4030_MICAMPL_EN; + avadc_ctl |= TWL4030_ADCL_EN; + } else { + anamicl &= ~TWL4030_MICAMPL_EN; + avadc_ctl &= ~TWL4030_ADCL_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + int value = 0; + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + switch (reg) { + case TWL4030_SUBMIC_EN: + value = 1; + break; + case TWL4030_AUXR_EN: + value = 2; + break; + default: + break; + } + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias, avadc_ctl; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 2: + anamicr |= TWL4030_AUXR_EN; + break; + default: + break; + } + + if (value != 0) { + anamicr |= TWL4030_MICAMPR_EN; + avadc_ctl |= TWL4030_ADCR_EN; + } else { + anamicr &= ~TWL4030_MICAMPR_EN; + avadc_ctl &= ~TWL4030_ADCR_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "None", + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "None", + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -378,6 +534,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for input amplifiers + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -420,6 +582,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, input_gain_tlv), + + /* Input source controls */ + SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 09865d9..a2065d4 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,7 +113,16 @@ #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ + +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + /* ANAMICL (0x05) Fields */ + #define TWL4030_CNCL_OFFSET_START 0x80 #define TWL4030_OFFSET_CNCL_SEL 0x60 #define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 @@ -127,10 +136,17 @@ #define TWL4030_MAINMIC_EN 0x01 /* ANAMICR (0x06) Fields */ + #define TWL4030_MICAMPR_EN 0x10 #define TWL4030_AUXR_EN 0x04 #define TWL4030_SUBMIC_EN 0x01 +/* AVADC_CTL (0x07) Fields */ + +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + /* AUDIO_IF (0x0E) Fields */ #define TWL4030_AIF_SLAVE_EN 0x80 -- 1.5.4.3

3 3

[alsa-devel] Gateway MX3042 Laptop. No sound
by Nigel Henry 03 Dec '08

03 Dec '08

I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but no sounds. The output from running alsa-info.sh is below. !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Tue Dec 2 16:30:06 CST 2008 !!Linux Distribution !!------------------ Ubuntu 8.04.1 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04.1" !!Kernel Information !!------------------ Kernel release: 2.6.24-22-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.16 Library version: Utilities version: 1.0.15 !!Loaded ALSA modules !!------------------- snd_intel8x0 !!Soundcards recognised by ALSA !!----------------------------- 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with AD1981B at irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1f.5 0401: 8086:24c5 (rev 03) Subsystem: 161f:203e !!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-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 !!Loaded sound module options !!-------------------------- !!Module: snd_intel8x0 ac97_clock : 0 ac97_quirk : <NULL> buggy_irq : N buggy_semaphore : N enable : N id : <NULL> index : -1 joystick : 0 spdif_aclink : 0 xbox : N !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Analog Devices AD1981B PCI Subsys Vendor: 0x161f PCI Subsys Device: 0x203e Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 16-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : 20dB [ 20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=10/11 SPDIF VRA PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz AD18XX configuration Unchained : 0x1000,0x0000,0x0000 Chained : 0x0000,0x0000,0x0000 0:00 = 0090 0:02 = 0000 0:04 = 0000 0:06 = 0000 0:08 = 0000 0:0a = 0000 0:0c = 001f 0:0e = 805f 0:10 = 0808 0:12 = 0606 0:14 = 0000 0:16 = 1f1f 0:18 = 0000 0:1a = 0505 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0605 0:2a = 0435 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2824 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 4000 0:5e = 0000 0:60 = 8080 0:62 = 0000 0:64 = 8000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0000 0:70 = 0000 0:72 = 1800 0:74 = 1001 0:76 = 2050 0:78 = 0000 0:7a = 0000 0:7c = 4144 0:7e = 5374 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 2008-12-02 11:09 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 24 2008-12-02 11:09 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 2008-12-02 14:27 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 25 2008-12-02 11:09 /dev/snd/pcmC0D1c crw-rw---- 1 root audio 116, 26 2008-12-02 11:09 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 27 2008-12-02 11:09 /dev/snd/pcmC0D3c crw-rw---- 1 root audio 116, 20 2008-12-02 11:09 /dev/snd/pcmC0D4p crw-rw---- 1 root audio 116, 1 2008-12-02 11:09 /dev/snd/seq crw-rw---- 1 root audio 116, 33 2008-12-02 11:09 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 1: Intel ICH - MIC ADC [Intel 82801DB-ICH4 - MIC ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 2: Intel ICH - MIC2 ADC [Intel 82801DB-ICH4 - MIC2 ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 3: Intel ICH - ADC2 [Intel 82801DB-ICH4 - ADC2] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [I82801DBICH4] Card hw:0 'I82801DBICH4'/'Intel 82801DB-ICH4 with AD1981B at irq 17' Mixer name : 'Analog Devices AD1981B' Components : 'AC97a:41445374' Controls : 34 Simple ctrls : 23 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 23 [74%] [0.00dB] [on] Capture [off] Front Right: Playback 23 [74%] [0.00dB] [on] Capture [off] Simple mixer control 'Line Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 25 [81%] [3.00dB] [on] Capture [off] Front Right: Playback 25 [81%] [3.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mic Boost ( 20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 3 [100%] Simple mixer control 'IEC958 Playback Source',0 Capabilities: enum Items: 'AC-Link' 'A/D Converter' Item0: 'AC-Link' Simple mixer control 'Aux',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [on] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [on] Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [on] Front Right: Capture 0 [0%] [0.00dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.I82801DBICH4 { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Master Playback Volume' value.0 31 value.1 31 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Headphone Playback Volume' value.0 31 value.1 31 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Master Mono Playback Volume' value 31 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value true } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Phone Playback Volume' value 0 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Mic Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost ( 20dB)' value true } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Line Playback Volume' value.0 23 value.1 23 } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'CD Playback Volume' value.0 25 value.1 25 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Aux Playback Switch' value.0 true value.1 true } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'PCM Playback Switch' value.0 true value.1 true } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'PCM Playback Volume' value.0 31 value.1 31 } control.20 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mix value.1 Mix } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.27 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.29 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 3 } control.30 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 AC-Link comment.item.1 'A/D Converter' iface MIXER name 'IEC958 Playback Source' value AC-Link } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Stereo Mic' value true } control.32 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Jack Sense' value true } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Jack Sense' value true } control.34 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat usb_storage libusual ipv6 af_packet rfcomm l2cap bluetooth rfkill_input i915 drm ppdev cpufreq_userspace cpufreq_stats cpufreq_powersave cpufreq_conservative cpufreq_ondemand freq_table dock container sbs sbshc iptable_filter ip_tables x_tables aes_i586 dm_crypt dm_mod sbp2 parport_pc lp parport joydev tifm_sd mmc_core pcmcia arc4 ecb blkcipher b43 rfkill mac80211 cfg80211 led_class input_polldev snd_intel8x0 snd_ac97_codec ac97_bus snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy video output evdev psmouse tifm_7xx1 serio_raw snd_seq_oss tifm_core yenta_socket rsrc_nonstatic snd_seq_midi pcmcia_core snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device battery ac button pcspkr snd soundcore iTCO_wdt iTCO_vendor_support snd_page_alloc intel_agp shpchp pci_hotplug agpgart ext3 jbd mbcache sg sr_mod cdrom sd_mod ata_generic usbhid hid b44 ata_piix ohci1394 pata_acpi ssb mii ieee1394 libata scsi_mod ehci_hcd uhci_hcd usbcore thermal processor fan fbcon tileblit font bitblit softcursor fuse Any help in getting Michael's sounds working, greatly appreciated. Nigel.

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[alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by scott＠waye.co.uk 03 Dec '08

03 Dec '08

From: root <scott(a)waye.co.uk> Signed-off-by: root <scott(a)waye.co.uk> diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c index 1360d54..b78c8f2 100644 --- a/pci/hda/patch_nvhdmi.c +++ b/pci/hda/patch_nvhdmi.c @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; -- 1.5.6

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[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------060204020609000202040409--

1 0

[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------070404090902050101040901--

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[alsa-devel] alsa-driver-1.0.18rc3 fails to compile against linux-2.6.27.7/arch/powerpc
by kordex - 03 Dec '08

03 Dec '08

make -C /lib/modules/2.6.27.7/sourceSUBDIRS=/root/alsa/alsa-driver-1.0.18rc3 CPP="gcc -E" CC="gcc" modules make[1]: Siirrytään hakemistoon "/usr/src/linux-2.6.27.7" CC [M] /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o In file included from /usr/src/linux-2.6.27.7/arch/powerpc/include/asm/system.h:528, from include/linux/list.h:7, from include/linux/module.h:9, from /root/alsa/alsa-driver-1.0.18rc3/include/adriver.h:26, from /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.c:1: include/asm-generic/cmpxchg-local.h: In function '__cmpxchg_local_generic': include/asm-generic/cmpxchg-local.h:23: error: implicit declaration of function 'local_irq_save' include/asm-generic/cmpxchg-local.h:44: error: implicit declaration of function 'local_irq_restore' make[3]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o] Virhe 1 make[2]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore] Virhe 2 make[1]: *** [_module_/root/alsa/alsa-driver-1.0.18rc3] Virhe 2 make[1]: Poistutaan hakemistosta "/usr/src/linux-2.6.27.7" make: *** [compile] Virhe 2 root@powerline:/root/alsa/alsa-driver-1.0.18rc3# root@powerline:/root/alsa/alsa-driver-1.0.18rc3# I used this configure line (compiling for emu 1212m): ./configure --with-cards=emu10k1 --with-sequencer=yes --with-oss=yes Gcc version: root@powerline:/root/alsa/alsa-driver-1.0.18rc3# gcc -v Using built-in specs. Target: powerpc-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr --enable-shared --with-system-zlib --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --enable-nls --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc --enable-mpfr --disable-softfloat --enable-secureplt --enable-targets=powerpc-linux,powerpc64-linux --with-cpu=default32 --with-long-double-128 --enable-checking=release --build=powerpc-linux-gnu --host=powerpc-linux-gnu --target=powerpc-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1)

1 0

[alsa-devel] Where is xfi driver now?
by Nickolay Hodyunya 02 Dec '08

02 Dec '08

Hello, I have updated to the latest git tree of sound-usntable kernel 1 day ago and now I can not find xfi drivers. Where are they go? -- Regards, Nickolay Hodyunya.

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[alsa-devel] [PATCH] ALSA: ac97 - Include ac97_codec.h for ac97_bus_type declaration
by Mark Brown 02 Dec '08

02 Dec '08

This fixes a sparse warning caused by the lack of a connection with the prototype for ac97_bus_type. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/ac97_bus.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 7fa37e1..a351dd0 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/string.h> +#include <sound/ac97_codec.h> /* * Let drivers decide whether they want to support given codec from their -- 1.5.6.5

2 1

[alsa-devel] [PATCH 1/1] intel hda: add model dell-m6 quirk for the Dell Studio 15 (1537)
by Andy Whitcroft 02 Dec '08

02 Dec '08

Sort out the headphone connections for the Dell Studio 15 (1537), by default there is no sound from this socket. This seems to have the same wiring as a dell-m6. Automatically detect and select that model. Signed-off-by: Andy Whitcroft <apw(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 243f8de..7760636 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1325,6 +1325,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M6), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f, + "unknown Dell", STAC_DELL_M6), {} /* terminator */ }; -- 1.6.0.4.911.gc990

2 2

[alsa-devel] [PATCH] ASoC: TWL4030: Add capture source selection and mic gain control
by Grazvydas Ignotas 02 Dec '08

02 Dec '08

The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds controls for capture source controls. When a microphone is selected, handlers automatically enable bias power (disable when deselected). The patch also adds microphone gain control. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/codecs/twl4030.c | 139 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 7 ++ 2 files changed, 146 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120..755a872 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -347,6 +347,130 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + + switch (value) { + case 0: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 1: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 2: + anamicl |= TWL4030_AUXL_EN; + break; + case 3: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + return -EINVAL; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + if (reg == TWL4030_SUBMIC_EN) + ucontrol->value.integer.value[0] = 0; + else if (reg == TWL4030_AUXR_EN) + ucontrol->value.integer.value[0] = 1; + + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + + switch (value) { + case 0: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 1: + anamicr |= TWL4030_AUXR_EN; + break; + default: + return -EINVAL; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -378,6 +502,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for microphone amplifier + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(microphone_gain_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -420,6 +550,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Microphone Boost", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, microphone_gain_tlv), + + /* Capture source controls */ + SOC_ENUM_EXT("Left Capture Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Capture Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 09865d9..8dae844 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,13 @@ #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + /* ANAMICL (0x05) Fields */ #define TWL4030_CNCL_OFFSET_START 0x80 #define TWL4030_OFFSET_CNCL_SEL 0x60 -- 1.5.4.3

3 5

[alsa-devel] ASoC: TWL4030 codec: Multiple DAC nameing
by Peter Ujfalusi 02 Dec '08

02 Dec '08

Hello, Now I have some working prototype for the TWL4030 codec DAPM playback routing. But before I send the patches I want to clear out something: Since the TWL has four DACs and I want to map all of them, what would be the correct SND_SOC_DAPM_DAC naming for them? At the moment I have these: /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", TWL4030_REG_AVDAC_CTL, 0, 0), SND_SOC_DAPM_DAC("DACL1", "Left Front Playback", TWL4030_REG_AVDAC_CTL, 1, 0), SND_SOC_DAPM_DAC("DACR2", "Right Rear Playback", TWL4030_REG_AVDAC_CTL, 2, 0), SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback", TWL4030_REG_AVDAC_CTL, 3, 0), Are these are OK, or should I just use "Right Playback" for both DACR1 and DACR2? Btw: the 'dynamic' routing and switching now works (no user visible power control for the DACs and APGAs) in a way, that if the output is configured to take the audio from one of the DAC, than the corresponding chain (APGA, DAC) will be powered on only. Thanks, Péter

2 1