June 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

(resending since I haven't gotten a reply yet) Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] snd-usb-caiaq: Fix compatibility with kernels < 2.6.19
by Sergey Vlasov 12 Jun '07

12 Jun '07

Kernels before 2.6.19 used a different signature for the urb->complete callback (an additional "struct pt_regs *regs" parameter was present), therefore snd-usb-caiaq needs some compatibility patches in alsa-driver, like snd-usb-audio. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- The snd_usb_caiaq_midi_output_done() part is somewhat hackish - the problem is that snd_usb_caiaq_midi_output_done() is declared in caiaq-midi.h, and the alsa-driver build system does not support patching of *.h files. The rest is basically copied from usbaudio.patch. Without the patch compiling for 2.6.18 produces warnings: .../usb/caiaq/caiaq-device.c: In function 'init_card': .../usb/caiaq/caiaq-device.c:316: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-device.c:321: warning: passing argument 6 of 'usb_fill_bulk_urb' from incompatible pointer type .../usb/caiaq/caiaq-audio.c: In function 'alloc_urbs': .../usb/caiaq/caiaq-audio.c:583: warning: assignment from incompatible pointer type --- alsa-driver-1.0.14/usb/caiaq/Makefile.alt-caiaq-api 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/usb/caiaq/Makefile 2007-06-11 19:50:39 +0400 @@ -5,6 +5,13 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf +clean-files := caiaq-audio.c caiaq-device.c + include $(SND_TOPDIR)/alsa-kernel/usb/caiaq/Makefile +EXTRA_CFLAGS += -I$(SND_TOPDIR)/alsa-kernel/usb/caiaq + include $(SND_TOPDIR)/Rules.make + +caiaq-audio.c: caiaq-audio.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-audio.c +caiaq-device.c: caiaq-device.patch $(SND_TOPDIR)/alsa-kernel/usb/caiaq/caiaq-device.c --- alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch.alt-caiaq-api 2007-06-11 19:18:40 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-audio.patch 2007-06-11 19:18:17 +0400 @@ -0,0 +1,26 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-audio.c 2007-06-11 16:29:28 +0400 ++++ caiaq-audio.c 2007-06-11 19:17:38 +0400 +@@ -464,7 +464,11 @@ static void fill_out_urb(struct snd_usb_ + check_for_elapsed_periods(dev, dev->sub_playback); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void read_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void read_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev; +@@ -518,7 +522,11 @@ static void read_completed(struct urb *u + usb_submit_urb(urb, GFP_ATOMIC); + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void write_completed(struct urb *urb, struct pt_regs *regs) ++#else + static void write_completed(struct urb *urb) ++#endif + { + struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaqdev *dev = info->dev; --- alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch.alt-caiaq-api 2007-06-11 19:55:17 +0400 +++ alsa-driver-1.0.14/usb/caiaq/caiaq-device.patch 2007-06-11 19:54:49 +0400 @@ -0,0 +1,31 @@ +--- ../../alsa-kernel/usb/caiaq/caiaq-device.c 2007-06-11 16:29:28 +0400 ++++ caiaq-device.c 2007-06-11 19:54:46 +0400 +@@ -100,7 +100,11 @@ static struct usb_device_id snd_usb_id_t + { /* terminator */ } + }; + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void usb_ep1_command_reply_dispatch (struct urb* urb, struct pt_regs *regs) ++#else + static void usb_ep1_command_reply_dispatch (struct urb* urb) ++#endif + { + int ret; + struct snd_usb_caiaqdev *dev = urb->context; +@@ -295,6 +299,16 @@ static struct snd_card* create_card(stru + return card; + } + ++#if !defined(OLD_USB) && !defined(CONFIG_SND_NEW_IRQ_HANDLER) ++static void snd_usb_caiaq_midi_output_done_wrapper(struct urb* urb, ++ struct pt_regs *regs) ++{ ++ snd_usb_caiaq_midi_output_done(urb); ++} ++#undef snd_usb_caiaq_midi_output_done ++#define snd_usb_caiaq_midi_output_done snd_usb_caiaq_midi_output_done_wrapper ++#endif ++ + static int init_card(struct snd_usb_caiaqdev *dev) + { + char *c;

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[alsa-devel] [PATCH] asihpi: Fix compatibility with 64-bit resource_size_t
by Sergey Vlasov 12 Jun '07

12 Jun '07

In recent kernels the resource_size_t type can be either 32-bit or 64-bit depending on config options. The proper way to printk such values is to cast them to unsigned long long. Signed-off-by: Sergey Vlasov <vsu(a)altlinux.ru> --- Without the patch compilation on 2.6.18 produces warnings: .../pci/asihpi/hpimod.c: In function 'adapter_probe': .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 4 has type 'resource_size_t' .../pci/asihpi/hpimod.c:460: warning: format '%x' expects type 'unsigned int', but argument 5 has type 'resource_size_t' --- alsa-driver-1.0.14/pci/asihpi/hpimod.c.alt-asihpi-warnings 2007-06-11 16:29:28 +0400 +++ alsa-driver-1.0.14/pci/asihpi/hpimod.c 2007-06-11 19:58:30 +0400 @@ -457,10 +457,10 @@ static int __devinit adapter_probe(struc // fill in HPI_PCI information from kernel provided information for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { - HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %x-%x\n", idx, + HPI_DEBUG_LOG4(DEBUG, "Resource %d %s %llx-%llx\n", idx, pci_dev->resource[idx].name, - pci_resource_start(pci_dev, idx), - pci_resource_end(pci_dev, idx)); + (unsigned long long)pci_resource_start(pci_dev, idx), + (unsigned long long)pci_resource_end(pci_dev, idx)); memlen = pci_resource_len(pci_dev, idx); if (memlen) {

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[alsa-devel] Shutdown() vector in ALSA
by Nobin Mathew 12 Jun '07

12 Jun '07

why no ALSA driver has shutdown() vector implemented? Is this vector needed to ensure the proper shutdown of the system? What all needs to be taken care if i want to implement shutdown(), except hardware shutdown. struct snd_shutdown_f_ops { struct file_operations f_ops; struct snd_shutdown_f_ops *next; }; what this operations means? defined in sound/core/init.c Is there any need to call these functions? snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]);

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Re: [alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Andrew Callan 12 Jun '07

12 Jun '07

I have been without sound using Fedora Core ever since the 2.6.16 or 2.6.17 kernel update (2.6.16 crashed during initialization, so I could not tell whether sound worked or not). It had worked well through the 2.6.15 Fedora kernels. I finally sat down tonight to work the problem as hard as I could and happened upon Eric and Tobin's timely exchange. First, simply adding probe_mask=1 to the snd-hda-intel options worked for me as well; I did not end up having to specify a specific model. Second, I too was seeing a LONG hang when trying to do: cat /proc/asound/card0/codec#0 but now with the probe_mask=1, I do not see this. If it helps, the alsa-info.sh output for my system is at http://pastebin.ca/496293 If it makes any difference, I'm not sure which model I had specified on the modprobe line at the time I captured this output. One thing I do notice is that after adding probe_mask=1, aplay -l does not show the Si3054 Modem which shows up in the alsa-info output. Thanks for the help you gave in the thread with Eric. Andrew

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[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 12 Jun '07

12 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Resent from the correct email address. Ignore previous email.

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[alsa-devel] snd_pcm_lib_malloc_pages()
by Timur Tabi 12 Jun '07

12 Jun '07

Liam Why do none of the ASoC drivers use snd_pcm_lib_malloc_pages()? They all allocate their DMA buffers by hand. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [ASoC] machine driver and channel configuration
by Matthieu CASTET 12 Jun '07

12 Jun '07

Hi, If the platform/cpu_dai driver and codec support capture and playback channels. But on the board only the playback link is connected. In this configuration, the codec and platform/cpu_dai driver shouldn't try to use the capture path (and the resource associated to this). Is there a way to do something like that ? Thanks Matthieu

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[alsa-devel] Questions on usb audio development
by LCID Fire 12 Jun '07

12 Jun '07

Hi. I know there is quite some documentation how to develop a pci driver for alsa - but since I'll try to get a usb device working I was wondering how to write a usb driver? Is it just the same? How to interface the various aspects of the device with regards to descriptor reading, ...

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[alsa-devel] hda-codec - Add HP Pavillion quirk to Realtek code.
by Tobin Davis 11 Jun '07

11 Jun '07

This patch adds the HP Pavillion a1589L desktop system to the ALC883 configuration table. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Patch - Revo7.1 PCM Playback Switch
by dustin＠seznam.cz 11 Jun '07

11 Jun '07

Hello Takashi, I have attached a patch to support mute for front channels of M-Audio Revolution 7.1 (the DAC AK4381 features a mute bit). Signed-off-by: Pavel Hofman <dustin at seznam.cz> Please could I ask you for your opinion on the following issue (so far no response): I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c). There is still a number of issues to solve but the 16 channel capture already works fine. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Best regards, Pavel.

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Re: [alsa-devel] [2/3] 2.6.22-rc4: known regressions v2
by Michal Piotrowski 09 Jun '07

09 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown IDE Subject : libata and legacy ide pcmcia failure References : http://lkml.org/lkml/2007/5/17/305 Submitter : Robert de Rooy <robert.de.rooy(a)gmail.com> Handled-By : Mark Lord <mlord(a)pobox.com> Status : problem is being debugged SATA/PATA Subject : 22-rc3 broke the CDROM in Dell notebook References : http://lkml.org/lkml/2007/5/27/63 Submitter : Gregor Jasny <gjasny(a)googlemail.com> Handled-By : Tejun Heo <htejun(a)gmail.com> Jeff Garzik <jeff(a)garzik.org> Caused-By : Tejun Heo <htejun(a)gmail.com> commit d4b2bab4f26345ea1803feb23ea92fbe3f6b77bc Status : problem is being debugged Sparc64 Subject : 2.6.22-rc broke X on Ultra5 References : http://lkml.org/lkml/2007/5/22/78 Submitter : Mikael Pettersson <mikpe(a)it.uu.se> Handled-By : David Miller <davem(a)davemloft.net> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] probing state of high definition audio card in windows / SigmaTel STAC9221 microphone problems
by Eran Guendelman 09 Jun '07

09 Jun '07

Hi, I am struggling to get the (external) microphone working on a SigmaTel STAC9221 A1 device on my Dell XPS M1210 laptop. I saw other emails indicating that this was still problematic, and the latest drivers from alsa 1.0.14 did not seem to fix the problem. So now that I've started reading and trying to understand the basics behind the hda codec in an effort to fix this problem, I was wondering if it was possible for me to boot into windows (I've got a dual-boot setup), get into a state where the microphone is working, and somehow probe the codec's state at that time (all of the pin configurations, etc.)... then theoretically, if I reproduce that configuration on the linux side, it should work (right?). So the question is, does anyone have a ready-made utility for Windows to do this probe of the high definition audio codec? I could probably try writing one myself, but thought I'd check if it already exists. Thanks. Please send responses by email too (not subscribed).

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[alsa-devel] Sound Devices USBPre
by Brian Phelps 08 Jun '07

08 Jun '07

Hey guys, I contacted Sound Devices at sounddevices.com about writing a Linux driver (I was going to head the project but would take any help I can get) for the USBPre, a 2ch-in 2ch-out USB preamp with highquality balanced XLR Mic inputs. They said that they would love to have a Linux Driver and will provide specs for me, I am still waiting on the info though. I have not asked if I can view their firmware sourcecode yet, and if it requires an NDA, I will have to work around that. What advice do you have for writing a driver for alsa? I have written some small kernel drivers for embedded Linux devices, a few audio programs in C, and I come from a DSP background, but I am not sure exactly where to start, besides the specs and firmware. Thanks for your help -Brian Phelps

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[alsa-devel] Callback trouble
by Juan Manuel Lopez Baio 08 Jun '07

08 Jun '07

Hello. I'm having some problems with the asynchronous usage of ALSA API, and I'm not sure if this usage is correct. What I have done so far is the following: I have an class that encapsulates the ALSA API, working in a multithreaded environment. I use alsa to capture audio and want to do so in an asynchronous and non-blocing way, so I defined a static method in my class that functions as callback, which receives a pointer to the instance of my class and calls on a member function, like this: void alsaDevice::myCallback(snd_async_handler_t *handler) //STATIC { alsaDevice *dev = (alsaDevice*)snd_async_handler_get_callback_private(handler); dev->triggerCondition(); } void alsaDevice::triggerCondition() { LogError(0,"Trigger en placa %d device %d",cn,dn); m_condMutex.lock(); m_noDataCondition.trigger(); m_condMutex.unlock(); } To init ALSA, I do this: int mode = SND_PCM_ASYNC | SND_PCM_NONBLOCK; if ((err=snd_pcm_open(&pcm_cap_handle, pcm_name.str().c_str(), SND_PCM_STREAM_CAPTURE, mode) < 0)) { LogError(0,"Opening ALSA PCM device %s: %s",pcm_name.str().c_str(),snd_strerror(err)); pcm_cap_handle=NULL; return false; } then set all the pertinent parameters, and finally if((asyncaddresult=snd_async_add_pcm_handler(&ahandler,pcm_cap_handle,alsaDevice::myCallback,this))<0) { LogError(0,"No se pudo registrar la callback; PCM Error: %i",asyncaddresult); } if ((asyncaddresult=snd_pcm_start(pcm_cap_handle))<0) LogError(0,"No se pudo start pcm; PCM Error: %i",asyncaddresult); Afterwards, the first time that a call to snd_pcm_readn returns EAGAIN I call wait() on a condition object with a 100 ms timeout, which is triggered by the callback, as seen before. The problem is that the callback never seems to get called (so shows the log), so I'm not really sure if the previous steps were right. Any light you can shed on this would be much appreciated. Thanks! juan

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[alsa-devel] snd-usb-audio: Add another QuickCam ID
by Ritesh Raj Sarraf 08 Jun '07

08 Jun '07

Hi, This patch adds another Logitech QuickCam USB ID for the device which is shipped with Dell's XPS M1210 Notebooks. Signed-off-by: Ritesh Raj Sarraf <rrs(a)researchut.com> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 374fbf6..c69a56f 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -70,6 +70,15 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08c6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, PS: Request you to CC me if there are any comments. I'm not subscribed to the list. Thanks, Ritesh -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention."

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[alsa-devel] [PATCH] soc/sh: let SND_SOC_PCM_SH7760 depend on SH_DMABRG
by Manuel Lauss 08 Jun '07

08 Jun '07

Let SND_SOC_PCM_SH7760 depend on SH_DMABRG rather than select'ing it. Gets rid of the Kbuild warning about SH_DMABRG being undefined on non-sh builds. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig~ 2007-06-07 11:09:30.000000000 +0200 +++ linux-2.6.22-rc4-mm2/sound/soc/sh/Kconfig 2007-06-07 11:09:30.000000000 +0200 @@ -2,8 +2,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC - select SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support.

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[alsa-devel] [PATCH] Add quirk for another MSI laptop to alc883_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding the SSID of another MSI laptop to the quirk list. Thanks, Daniel Chen

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[alsa-devel] [ASoC]one Small Suggestion
by Nobin Mathew 08 Jun '07

08 Jun '07

Liam, Thank you for your support, my work is almost complete now, driver is working perfectly now. I have one suggestion, I modified the wm9713.c codec driver in probe function , so that if(socdev->machine->dai_link->cpu_dai->private_data) codec->ac97->bus->private_data = socdev->machine->dai_link->cpu_dai->private_data; Here i am assigning the controller private data. so that my following ac97 read and write to access codec registers become good. static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct platform_controller *controller = ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; u32 v; Is there a way to accomodate this in existing ASoC design? Or a better way to do this?

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[alsa-devel] [RFC] [PATCH 1/2] ASoC sound support for SMDK2440 boards: UDA1341 driver
by Zoltan Devai 08 Jun '07

08 Jun '07

ASoC Support for the UDA1341 codec. Please note that until there is a final L3-bus driver, this patch won't work. --- --- /dev/null +++ /sound/soc/codecs/uda1341.c @@ -0,0 +1,519 @@ +/* + * uda1341.c -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * 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. + */ + +#include <linux/module.h> +#include <linux/l3.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "../s3c24xx/s3c24xx-i2s.h" +#include "uda1341.h" + +#define UDA1341_DEBUG 0 + +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "UDA1341: %s:" format "\n" ,__FUNCTION__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE ) + +struct uda1341_priv { + int sysclk; + int dai_fmt; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %d", reg); + + if (reg >= UDA1341_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + DBG("reg: %02X", reg); + + if (reg >= UDA1341_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda1341 registers + * + */ +static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + int addr; + u8 data = value; + + DBG("reg: %02X, value:%02X", reg, value); + + if (reg >= UDA1341_REGS_NUM) { + DBG("Unkown register: reg: %d", reg); + return -EINVAL; + } + + uda1341_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA1341_STATUS0: + case UDA1341_STATUS1: + addr = UDA1341_STATUS_ADDR; + break; + case UDA1341_DATA000: + case UDA1341_DATA001: + case UDA1341_DATA010: + addr = UDA1341_DATA0_ADDR; + break; + case UDA1341_DATA1: + addr = UDA1341_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA1341_EXTADDR_PREFIX); + + ret = codec->hw_write_l3(codec->control_data, + UDA1341_DATA0_ADDR, (char*)&addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA1341_DATA0_ADDR; + data = (value | UDA1341_EXTDATA_PREFIX); + break; + } + + ret = codec->hw_write_l3(codec->control_data, addr, (char*)&data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda1341_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6)); +} + +static int uda1341_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010); + + DBG("mute: %d", mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda1341_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda1341->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + return -EINVAL; + break; + } + + DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt, + params_format(params)); + + + /* set DAI format and word length */ + switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch(params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + return -EINVAL; + } + + uda1341_write(codec, UDA1341_STATUS0, hw_params); + + return 0; +} + +static int uda1341_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + we'll error out on set_hw_params if it's not OK */ + if ( (freq >= (256 * 8000)) && (freq <= (512 * 48000)) ) { + uda1341->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int uda1341_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("fmt: %08X", fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1341->dai_fmt = fmt; + + return 0; +} + +static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{ + u8 reg; + + DBG("event: %08X", event); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* ADC, DAC on */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg | 0x03); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* mute, ADC, DAC power off */ + reg = codec->read(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03)); + break; + } + codec->dapm_state = event; + return 0; +} + +static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1", "Minimum2", "Maximum"}; +static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2", "Input 2", "Digital mixer"}; + +static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0), + +SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]), + +/* This should be an ext control with own handler, if one wants + to set the values in 0.5dB steps instead of 3dB */ +SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +}; + +static int uda1341_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) { + if ((err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1341_snd_controls[i],codec, NULL))) < 0) + return err; + } + + return 0; +} + +struct snd_soc_codec_dai uda1341_dai = { + .name = "UDA1341", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* pcm operations */ + .ops = { + .hw_params = uda1341_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda1341_mute, + .set_sysclk = uda1341_set_dai_sysclk, + .set_fmt = uda1341_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda1341_dai); + + +static int uda1341_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1341_priv *uda1341; + void * codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + + printk(KERN_INFO "UDA1341 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL); + if (uda1341 == NULL) + goto priv_err; + + codec->private_data = uda1341; + + codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg), \ + GFP_KERNEL); + + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda1341_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA1341"; + codec->owner = THIS_MODULE; + codec->dai = &uda1341_dai; + codec->num_dai = 1; + codec->read = uda1341_read_reg_cache; + codec->write = uda1341_write; + codec->dapm_event = uda1341_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA1341 SoC codec: \ + Unable to find an L3 bus adapter\n"); + ret = -ENODEV; + goto pcm_err; + } + + codec->control_data = codec_setup_data; + codec->hw_write_l3 = (hw_write_l3_t)l3_write; + + uda1341_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda1341_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register controls\n"); + goto pcm_err; + } + + //uda1341_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + return 0; +} + +static int uda1341_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++) { + codec->write(codec, i, *cache++); + } + uda1341_dapm_event(codec, codec->suspend_dapm_state); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda1341 = { + .probe = uda1341_soc_probe, + .remove = uda1341_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda1341_soc_suspend, + .resume = uda1341_soc_resume, +#endif /* CONFIG_PM */ +}; + +EXPORT_SYMBOL(soc_codec_dev_uda1341); + +MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai"); +MODULE_LICENSE("GPL"); --- /dev/null +++ /sound/soc/codecs/uda1341.h @@ -0,0 +1,47 @@ +/* + * uda1341.h -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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. + */ + +#ifndef _UDA1341_H +#define _UDA1341_H + +#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2) + +#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0 + +/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13 + +#define UDA1341_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_codec_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341; + +#endif /* _UDA1341_H */

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[alsa-devel] [PATCH] Add SSID for Dell XPS400 to stac922x_cfg_tbl[]
by Daniel T. Chen 08 Jun '07

08 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip adding a quirk entry for a Dell XPS400. Thanks, Daniel Chen

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[alsa-devel] Testing/forcing XRUN
by Eliot Blennerhassett 08 Jun '07

08 Jun '07

Hi, I want to test my driver behaviour when an xrun happens. I have tried using aplay/arecord, and suspending (ctrl-Z) them while playing/recording but this doesn't have the desired effect. (I don't see my driver code detecting an xrun). Instead I see a trigger-stop command to the driver. The apps do report an overrun when they are restarted though. Can anyone answer these: 1) Is this a valid way to simulate an xrun due to application not keeping up? 1.1) If not, what is? Do I have to write an app that deliberately starves the streams? 2) Where is the xrun being detected if my driver doesnt see it? Where does the trigger-stop command come from when I suspend the app? thanks Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] Audio driver for omap 24xx.
by Pharaoh . 07 Jun '07

07 Jun '07

Hi I am referring the EAC driver for omap 24xx which is used by n800 board. The EAC 24xx driver doesn't look complete since it has only hardware init sequences and very basic ALSA related functionalities i.e. register the sound card etc. Is omap 24xx eac driver still in intermediate stage?Or it is complete but not yet integrated in to omap git tree? In later case, where can I find the complete eac driver? Also, omap 24xx driver doesn't seem to follow the framework/structure of other omap alsa drivers in sound/arm/omap/ , any particular reason? Please enlighten. I know this might be little off topic here, but there are only two places where I can ask this question, so I am trying both. -pharaoh.

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[alsa-devel] Surround sound problem with CS46xx
by Vasilis Manwlopoulos 06 Jun '07

06 Jun '07

Hi to everybody, I am kind of new in linux and to alsa and I am facing a problem with the setup for 5.1 surround sound with my sound card Hercules Fortissimo III, recognized having the chip cs46xx. I have managed to enable surround sound by creating a .asounrc file with the followings pcm.!default { type plug slave.pcm "surround51" slave.channels 6 route_policy duplicate } The problem is that I am facing weird cpu load spikes (sometimes isn't just spikes, but just 99% usage for a long time!) even when just playing a normal mp3 file at 128Kbps. This is happening when playing with various players (totem,rhythbox,vlc). Also the sound has cracklings or there is a delay between the front and the rear speakers. i have send this email to alsa-users too, but since there wasn't anyone to propose a workaround I was adviced to also send it to alsa-devel. I am using the distro ubuntu 7.04 with the default alsa libraries version 1.0.13ubuntu5. So, if anyone has any idea plz give me a hint, cause I am stuck with it about two weeks now, without having any clue to proceed with!!! PS. If any other information or command output is necessary to help me, tell me so.

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[alsa-devel] [ASoC]Changing the order of trigger in soc_pcm_trigger
by Nobin Mathew 06 Jun '07

06 Jun '07

Liam, I have one suggestion. Can we use this if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } Instead of if (rtd->codec_dai->ops.trigger) { ret = rtd->codec_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } if (platform->pcm_ops->trigger) { ret = platform->pcm_ops->trigger(substream, cmd); if (ret < 0) return ret; } if (rtd->cpu_dai->ops.trigger) { ret = rtd->cpu_dai->ops.trigger(substream, cmd); if (ret < 0) return ret; } That means trigger the Cpu before pcm. Because most of the controllers needs to be initialized and triggered before external DMA is triggered. Correct me if i am going wrong. This was causing so many problems in my hardware. Nobin Mathew

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